ACM: Remove functions related to DTMF

webrtc/modules/audio_coding/main/interface/audio_coding_module.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  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.
  */
 
@@ skipping 600 matching lines @@
 
   //
   // The shortest latency, in milliseconds, required by jitter buffer. This
   // is computed based on inter-arrival times and playout mode of NetEq. The
   // actual delay is the maximum of least-required-delay and the minimum-delay
   // specified by SetMinumumPlayoutDelay() API.
   //
   virtual int LeastRequiredDelayMs() const = 0;
 
   ///////////////////////////////////////////////////////////////////////////
-  // int32_t SetDtmfPlayoutStatus()
-  // Configure DTMF playout, i.e. whether out-of-band
-  // DTMF tones are played or not.
-  //
-  // Input:
-  //   -enable             : if true to enable playout out-of-band DTMF tones,
-  //                         false to disable.
-  //
-  // Return value:
-  //   -1 if the method fails, e.g. DTMF playout is not supported.
-  //    0 if the status is set successfully.
-  //
-  virtual int32_t SetDtmfPlayoutStatus(const bool enable) = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
-  // bool DtmfPlayoutStatus()
-  // Get Dtmf playout status.
-  //
-  // Return value:
-  //   true if out-of-band Dtmf tones are played,
-  //   false if playout of Dtmf tones is disabled.
-  //
-  virtual bool DtmfPlayoutStatus() const = 0;
-
-  ///////////////////////////////////////////////////////////////////////////
   // int32_t PlayoutTimestamp()
   // The send timestamp of an RTP packet is associated with the decoded
   // audio of the packet in question. This function returns the timestamp of
   // the latest audio obtained by calling PlayoutData10ms().
   //
   // Input:
   //   -timestamp          : a reference to a uint32_t to receive the
   //                         timestamp.
   // Return value:
   //    0 if the output is a correct timestamp.
@@ skipping 200 matching lines @@
       old_config.id = 0;
       old_config.neteq_config = neteq_config;
       old_config.clock = clock;
       return old_config;
     }
 
     NetEq::Config neteq_config;
     Clock* clock;
     AudioPacketizationCallback* transport;
     ACMVADCallback* vad_callback;
-    bool play_dtmf;
     int initial_playout_delay_ms;
     int playout_channels;
     int playout_frequency_hz;
   };
 
   static AudioCoding* Create(const Config& config);
   virtual ~AudioCoding() {};
 
   // Registers a codec, specified by |send_codec|, as sending codec.
   // This API can be called multiple times. The last codec registered overwrites
@@ skipping 134 matching lines @@
   virtual std::vector<uint16_t> GetNackList(int round_trip_time_ms) const = 0;
 
   // Returns the timing statistics for calls to Get10MsAudio.
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H_