NetEq: Add codec name and RTP timestamp rate to DecoderInfo

webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc

 /*
  *  Copyright (c) 2014 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 126 matching lines @@
       ASSERT_EQ(0, acm_->RegisterReceiveCodec(my_codec_param))
           << "Couldn't register receive codec.\n";
     }
   }
 }
 
 int AcmReceiveTestOldApi::RegisterExternalReceiveCodec(
     int rtp_payload_type,
     AudioDecoder* external_decoder,
     int sample_rate_hz,
-    int num_channels) {
+    int num_channels,
+    const std::string& name) {
   return acm_->RegisterExternalReceiveCodec(rtp_payload_type, external_decoder,
-                                            sample_rate_hz, num_channels);
+                                            sample_rate_hz, num_channels, name);
 }
 
 void AcmReceiveTestOldApi::Run() {
   for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
        packet.reset(packet_source_->NextPacket())) {
     // Pull audio until time to insert packet.
     while (clock_.TimeInMilliseconds() < packet->time_ms()) {
       AudioFrame output_frame;
       EXPECT_EQ(0, acm_->PlayoutData10Ms(output_freq_hz_, &output_frame));
       EXPECT_EQ(output_freq_hz_, output_frame.sample_rate_hz_);
@@ skipping 52 matching lines @@
   if (clock_.TimeInMilliseconds() >= last_toggle_time_ms_ + toggle_period_ms_) {
     output_freq_hz_ = (output_freq_hz_ == output_freq_hz_1_)
                           ? output_freq_hz_2_
                           : output_freq_hz_1_;
     last_toggle_time_ms_ = clock_.TimeInMilliseconds();
   }
 }
 
 }  // namespace test
 }  // namespace webrtc