Remove old methods in AudioTransport, make it pass a gn build

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 1200 matching lines @@
   // AudioSource::Sink implementation.
   // This method is called on the audio thread.
   void OnData(const void* audio_data,
               int bits_per_sample,
               int sample_rate,
               size_t number_of_channels,
               size_t number_of_frames) override {
     RTC_DCHECK(!worker_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(audio_capture_thread_checker_.CalledOnValidThread());
     RTC_DCHECK(voe_audio_transport_);
-    voe_audio_transport_->OnData(config_.voe_channel_id,
-                                 audio_data,
-                                 bits_per_sample,
-                                 sample_rate,
-                                 number_of_channels,
-                                 number_of_frames);
+    voe_audio_transport_->PushCaptureData(config_.voe_channel_id, audio_data,
+                                          bits_per_sample, sample_rate,
+                                          number_of_channels, number_of_frames);
   }
 
   // Callback from the |source_| when it is going away. In case Start() has
   // never been called, this callback won't be triggered.
   void OnClose() override {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
     // Set |source_| to nullptr to make sure no more callback will get into
     // the source.
     source_ = nullptr;
     UpdateSendState();
@@ skipping 1372 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE