NetEq: Add codec name and RTP timestamp rate to DecoderInfo

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.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 274 matching lines @@
     // Previous packet was not received.
     uniform_loss_model_->set_loss_rate(prob_trans_11_);
     return lost_last_ = uniform_loss_model_->Lost();
   } else {
     uniform_loss_model_->set_loss_rate(prob_trans_01_);
     return lost_last_ = uniform_loss_model_->Lost();
   }
 }
 
 void NetEqQualityTest::SetUp() {
-  ASSERT_EQ(0, neteq_->RegisterPayloadType(decoder_type_, kPayloadType));
+  ASSERT_EQ(0,
+            neteq_->RegisterPayloadType(decoder_type_, "noname", kPayloadType));
   rtp_generator_->set_drift_factor(drift_factor_);
 
   int units = block_duration_ms_ / kPacketLossTimeUnitMs;
   switch (FLAGS_random_loss_mode) {
     case 1: {
       // |unit_loss_rate| is the packet loss rate for each unit time interval
       // (kPacketLossTimeUnitMs). Since a packet loss event is generated if any
       // of |block_duration_ms_ / kPacketLossTimeUnitMs| unit time intervals of
       // a full packet duration is drawn with a loss, |unit_loss_rate| fulfills
       // (1 - unit_loss_rate) ^ (block_duration_ms_ / kPacketLossTimeUnitMs) ==
@@ skipping 121 matching lines @@
     }
   }
   Log() << "Average bit rate was "
         << 8.0f * total_payload_size_bytes_ / FLAGS_runtime_ms
         << " kbps"
         << std::endl;
 }
 
 }  // namespace test
 }  // namespace webrtc