Moved call.h and most of api/call/* into call/

webrtc/call/audio_send_stream.cc

 /*
  *  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.
  */
 
-#include "webrtc/api/call/audio_send_stream.h"
+#include "webrtc/call/audio_send_stream.h"
 
 #include <string>
 
 namespace {
 
 std::string ToString(const webrtc::CodecInst& codec_inst) {
   std::stringstream ss;
   ss << "{pltype: " << codec_inst.pltype;
   ss << ", plname: \"" << codec_inst.plname << "\"";
   ss << ", plfreq: " << codec_inst.plfreq;
@@ skipping 78 matching lines @@
       enable_opus_dtx == rhs.enable_opus_dtx &&
       opus_max_playback_rate == rhs.opus_max_playback_rate &&
       cng_payload_type == rhs.cng_payload_type &&
       cng_plfreq == rhs.cng_plfreq && max_ptime_ms == rhs.max_ptime_ms &&
       min_ptime_ms == rhs.min_ptime_ms && codec_inst == rhs.codec_inst) {
     return true;
   }
   return false;
 }
 }  // namespace webrtc