Multi frequency DTMF support - sender side

webrtc/audio/audio_send_stream.h

 /*
  *  Copyright (c) 2015 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 25 matching lines @@
                   const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
                   rtc::TaskQueue* worker_queue,
                   CongestionController* congestion_controller,
                   BitrateAllocator* bitrate_allocator,
                   RtcEventLog* event_log);
   ~AudioSendStream() override;
 
   // webrtc::AudioSendStream implementation.
   void Start() override;
   void Stop() override;
-  bool SendTelephoneEvent(int payload_type, int event,
+  bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
                           int duration_ms) override;
   void SetMuted(bool muted) override;
   webrtc::AudioSendStream::Stats GetStats() const override;
 
   void SignalNetworkState(NetworkState state);
   bool DeliverRtcp(const uint8_t* packet, size_t length);
 
   // Implements BitrateAllocatorObserver.
   uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
                             uint8_t fraction_loss,
@@ skipping 14 matching lines @@
   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   BitrateAllocator* const bitrate_allocator_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };
 }  // namespace internal
 }  // namespace webrtc
 
 #endif  // WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_