Refactor WVoE DTMF handling #2

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 127 matching lines @@
   bool experimental_ns_enabled() {
     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::VoEBase, public webrtc::VoECodec,
       public webrtc::VoEHardware,
       public webrtc::VoENetwork, public webrtc::VoERTP_RTCP,
       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),
           vad(false),
           codec_fec(false),
           max_encoding_bandwidth(0),
           opus_dtx(false),
           red(false),
           nack(false),
           cn8_type(13),
           cn16_type(105),
-          dtmf_type(106),
           red_type(117),
           nack_max_packets(0),
           send_ssrc(0),
           associate_send_channel(-1),
           recv_codecs(),
           neteq_capacity(-1),
           neteq_fast_accelerate(false) {
       memset(&send_codec, 0, sizeof(send_codec));
     }
     bool external_transport;
     bool send;
     bool playout;
     float volume_scale;
     bool vad;
     bool codec_fec;
     int max_encoding_bandwidth;
     bool opus_dtx;
     bool red;
     bool nack;
     int cn8_type;
     int cn16_type;
-    int dtmf_type;
     int red_type;
     int nack_max_packets;
     uint32_t send_ssrc;
     int associate_send_channel;
-    DtmfInfo dtmf_info;
     std::vector<webrtc::CodecInst> recv_codecs;
     webrtc::CodecInst send_codec;
     webrtc::PacketTime last_rtp_packet_time;
     std::list<std::string> packets;
     int neteq_capacity;
     bool neteq_fast_accelerate;
   };
 
   FakeWebRtcVoiceEngine()
       : inited_(false),
@@ skipping 60 matching lines @@
   }
   const webrtc::PacketTime& GetLastRtpPacketTime(int channel) {
     RTC_DCHECK(channels_.find(channel) != channels_.end());
     return channels_[channel]->last_rtp_packet_time;
   }
   int GetSendCNPayloadType(int channel, bool wideband) {
     return (wideband) ?
         channels_[channel]->cn16_type :
         channels_[channel]->cn8_type;
   }
-  int GetSendTelephoneEventPayloadType(int channel) {
-    return channels_[channel]->dtmf_type;
-  }
   int GetSendREDPayloadType(int channel) {
     return channels_[channel]->red_type;
   }
   bool CheckPacket(int channel, const void* data, size_t len) {
     bool result = !CheckNoPacket(channel);
     if (result) {
       std::string packet = channels_[channel]->packets.front();
       result = (packet == std::string(static_cast<const char*>(data), len));
       channels_[channel]->packets.pop_front();
     }
@@ skipping 248 matching lines @@
   WEBRTC_FUNC(SetOpusDtx, (int channel, bool enable_dtx)) {
     WEBRTC_CHECK_CHANNEL(channel);
     if (_stricmp(channels_[channel]->send_codec.plname, "opus") != 0) {
       // Return -1 if current send codec is not Opus.
       return -1;
     }
     channels_[channel]->opus_dtx = enable_dtx;
     return 0;
   }
 
-  // webrtc::VoEDtmf
-  WEBRTC_FUNC(SendTelephoneEvent, (int channel, int event_code,
-      bool out_of_band = true, int length_ms = 160, int attenuation_db = 10)) {
-    channels_[channel]->dtmf_info.dtmf_event_code = event_code;
-    channels_[channel]->dtmf_info.dtmf_out_of_band = out_of_band;
-    channels_[channel]->dtmf_info.dtmf_length_ms = length_ms;
-    return 0;
-  }
-  WEBRTC_FUNC(SetSendTelephoneEventPayloadType,
-      (int channel, unsigned char type)) {
-    channels_[channel]->dtmf_type = type;
-    return 0;
-  };
-  WEBRTC_STUB(GetSendTelephoneEventPayloadType,
-      (int channel, unsigned char& type));
-  WEBRTC_STUB(SetDtmfFeedbackStatus, (bool enable, bool directFeedback));
-  WEBRTC_STUB(GetDtmfFeedbackStatus, (bool& enabled, bool& directFeedback));
-  WEBRTC_STUB(PlayDtmfTone,
-      (int event_code, int length_ms = 200, int attenuation_db = 10));
-
   // 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 239 matching lines @@
   }
   bool IsHighPassFilterEnabled() {
     return highpass_filter_enabled_;
   }
   bool IsStereoChannelSwappingEnabled() {
     return stereo_swapping_enabled_;
   }
   void EnableStereoChannelSwapping(bool enable) {
     stereo_swapping_enabled_ = enable;
   }
-  bool WasSendTelephoneEventCalled(int channel, int event_code, int length_ms) {
-    return (channels_[channel]->dtmf_info.dtmf_event_code == event_code &&
-            channels_[channel]->dtmf_info.dtmf_out_of_band == true &&
-            channels_[channel]->dtmf_info.dtmf_length_ms == length_ms);
-  }
-  bool WasPlayDtmfToneCalled(int event_code, int length_ms) {
-    return (dtmf_info_.dtmf_event_code == event_code &&
-            dtmf_info_.dtmf_length_ms == length_ms);
-  }
   int GetNetEqCapacity() const {
     auto ch = channels_.find(last_channel_);
     ASSERT(ch != channels_.end());
     return ch->second->neteq_capacity;
   }
   bool GetNetEqFastAccelerate() const {
     auto ch = channels_.find(last_channel_);
     ASSERT(ch != channels_.end());
     return ch->second->neteq_fast_accelerate;
   }
@@ skipping 50 matching lines @@
   webrtc::EcModes ec_mode_;
   webrtc::AecmModes aecm_mode_;
   webrtc::NsModes ns_mode_;
   webrtc::AgcModes agc_mode_;
   webrtc::AgcConfig agc_config_;
   webrtc::VoiceEngineObserver* observer_;
   int playout_fail_channel_;
   int send_fail_channel_;
   int recording_sample_rate_;
   int playout_sample_rate_;
-  DtmfInfo dtmf_info_;
   FakeAudioProcessing audio_processing_;
 };
 
 }  // namespace cricket
 
 #endif  // TALK_SESSION_PHONE_FAKEWEBRTCVOICEENGINE_H_