Remove various IDs

modules/audio_coding/test/insert_packet_with_timing.cc

 /*
  *  Copyright (c) 2013 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 43 matching lines @@
 const int32_t kPacketPushedIn = 0x00000001 << 1;
 const int kPlayoutPeriodMs = 10;
 
 namespace webrtc {
 
 class InsertPacketWithTiming {
  public:
   InsertPacketWithTiming()
       : sender_clock_(new SimulatedClock(0)),
         receiver_clock_(new SimulatedClock(0)),
-        send_acm_(AudioCodingModule::Create(0, sender_clock_)),
-        receive_acm_(AudioCodingModule::Create(0, receiver_clock_)),
+        send_acm_(AudioCodingModule::Create(sender_clock_)),
+        receive_acm_(AudioCodingModule::Create(receiver_clock_)),
         channel_(new Channel),
         seq_num_fid_(fopen(FLAG_seq_num, "rt")),
         send_ts_fid_(fopen(FLAG_send_ts, "rt")),
         receive_ts_fid_(fopen(FLAG_receive_ts, "rt")),
         pcm_out_fid_(fopen(FLAG_output, "wb")),
         samples_in_1ms_(48),
         num_10ms_in_codec_frame_(2),  // Typical 20 ms frames.
         time_to_insert_packet_ms_(3),  // An arbitrary offset on pushing packet.
         next_receive_ts_(0),
         time_to_playout_audio_ms_(kPlayoutPeriodMs),
@@ skipping 237 matching lines @@
       if (delay_log != NULL) {
         fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms);
       }
     }
   }
   std::cout << std::endl;
   test.TearDown();
   if (delay_log != NULL)
     fclose(delay_log);
 }