Remove VoEFile from VoeWrapper and the remaining places in libjingle where it was being used.

talk/media/webrtc/fakewebrtcvoiceengine.h

 /*
  * libjingle
  * Copyright 2010 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,
@@ skipping 169 matching lines @@
     return experimental_ns_enabled_;
   }
 
  private:
   bool experimental_ns_enabled_;
 };
 
 class FakeWebRtcVoiceEngine
     : public webrtc::VoEAudioProcessing,
       public webrtc::VoEBase, public webrtc::VoECodec, public webrtc::VoEDtmf,
-      public webrtc::VoEFile, public webrtc::VoEHardware,
+      public webrtc::VoEHardware,
       public webrtc::VoEExternalMedia, public webrtc::VoENetEqStats,
       public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
       public webrtc::VoEVideoSync, public webrtc::VoEVolumeControl {
  public:
   struct DtmfInfo {
     DtmfInfo()
       : dtmf_event_code(-1),
         dtmf_out_of_band(false),
         dtmf_length_ms(-1) {}
     int dtmf_event_code;
     bool dtmf_out_of_band;
     int dtmf_length_ms;
   };
   struct Channel {
     explicit Channel()
         : external_transport(false),
           send(false),
           playout(false),
           volume_scale(1.0),
           volume_pan_left(1.0),
           volume_pan_right(1.0),
-          file(false),
           vad(false),
           codec_fec(false),
           max_encoding_bandwidth(0),
           opus_dtx(false),
           red(false),
           nack(false),
           media_processor_registered(false),
           rx_agc_enabled(false),
           rx_agc_mode(webrtc::kAgcDefault),
           cn8_type(13),
@@ skipping 11 matching lines @@
           neteq_fast_accelerate(false) {
       memset(&send_codec, 0, sizeof(send_codec));
       memset(&rx_agc_config, 0, sizeof(rx_agc_config));
     }
     bool external_transport;
     bool send;
     bool playout;
     float volume_scale;
     float volume_pan_left;
     float volume_pan_right;
-    bool file;
     bool vad;
     bool codec_fec;
     int max_encoding_bandwidth;
     bool opus_dtx;
     bool red;
     bool nack;
     bool media_processor_registered;
     bool rx_agc_enabled;
     webrtc::AgcModes rx_agc_mode;
     webrtc::AgcConfig rx_agc_config;
@@ skipping 461 matching lines @@
   WEBRTC_STUB(SetDtmfFeedbackStatus, (bool enable, bool directFeedback));
   WEBRTC_STUB(GetDtmfFeedbackStatus, (bool& enabled, bool& directFeedback));
 
   WEBRTC_FUNC(PlayDtmfTone,
       (int event_code, int length_ms = 200, int attenuation_db = 10)) {
     dtmf_info_.dtmf_event_code = event_code;
     dtmf_info_.dtmf_length_ms = length_ms;
     return 0;
   }
 
-  // webrtc::VoEFile
-  WEBRTC_FUNC(StartPlayingFileLocally, (int channel, const char* fileNameUTF8,
-                                        bool loop, webrtc::FileFormats format,
-                                        float volumeScaling, int startPointMs,
-                                        int stopPointMs)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->file = true;
-    return 0;
-  }
-  WEBRTC_FUNC(StartPlayingFileLocally, (int channel, webrtc::InStream* stream,
-                                        webrtc::FileFormats format,
-                                        float volumeScaling, int startPointMs,
-                                        int stopPointMs)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->file = true;
-    return 0;
-  }
-  WEBRTC_FUNC(StopPlayingFileLocally, (int channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    channels_[channel]->file = false;
-    return 0;
-  }
-  WEBRTC_FUNC(IsPlayingFileLocally, (int channel)) {
-    WEBRTC_CHECK_CHANNEL(channel);
-    return (channels_[channel]->file) ? 1 : 0;
-  }
-  WEBRTC_STUB(StartPlayingFileAsMicrophone, (int channel,
-                                             const char* fileNameUTF8,
-                                             bool loop,
-                                             bool mixWithMicrophone,
-                                             webrtc::FileFormats format,
-                                             float volumeScaling));
-  WEBRTC_STUB(StartPlayingFileAsMicrophone, (int channel,
-                                             webrtc::InStream* stream,
-                                             bool mixWithMicrophone,
-                                             webrtc::FileFormats format,
-                                             float volumeScaling));
-  WEBRTC_STUB(StopPlayingFileAsMicrophone, (int channel));
-  WEBRTC_STUB(IsPlayingFileAsMicrophone, (int channel));
-  WEBRTC_STUB(StartRecordingPlayout, (int channel, const char* fileNameUTF8,
-                                      webrtc::CodecInst* compression,
-                                      int maxSizeBytes));
-  WEBRTC_STUB(StartRecordingPlayout, (int channel, webrtc::OutStream* stream,
-                                      webrtc::CodecInst* compression));
-  WEBRTC_STUB(StopRecordingPlayout, (int channel));
-  WEBRTC_FUNC(StartRecordingMicrophone, (const char* fileNameUTF8,
-                                         webrtc::CodecInst* compression,
-                                         int maxSizeBytes)) {
-    return 0;
-  }
-  WEBRTC_FUNC(StartRecordingMicrophone, (webrtc::OutStream* stream,
-                                         webrtc::CodecInst* compression)) {
-    return 0;
-  }
-  WEBRTC_FUNC(StopRecordingMicrophone, ()) {
-    return 0;
-  }
-
   // webrtc::VoEHardware
   WEBRTC_FUNC(GetNumOfRecordingDevices, (int& num)) {
     return GetNumDevices(num);
   }
   WEBRTC_FUNC(GetNumOfPlayoutDevices, (int& num)) {
     return GetNumDevices(num);
   }
   WEBRTC_FUNC(GetRecordingDeviceName, (int i, char* name, char* guid)) {
     return GetDeviceName(i, name, guid);
   }
@@ skipping 467 matching lines @@
   DtmfInfo dtmf_info_;
   webrtc::VoEMediaProcess* media_processor_;
   FakeAudioProcessing audio_processing_;
 };
 
 #undef WEBRTC_CHECK_HEADER_EXTENSION_ID
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_