NetEq: Add codec name and RTP timestamp rate to DecoderInfo

webrtc/modules/audio_coding/neteq/tools/neteq_performance_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 16 matching lines @@
 namespace test {
 
 int64_t NetEqPerformanceTest::Run(int runtime_ms,
                                   int lossrate,
                                   double drift_factor) {
   const std::string kInputFileName =
       webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
   const int kSampRateHz = 32000;
   const webrtc::NetEqDecoder kDecoderType =
       webrtc::NetEqDecoder::kDecoderPCM16Bswb32kHz;
+  const std::string kDecoderName = "pcm16-swb32";
   const int kPayloadType = 95;
 
   // Initialize NetEq instance.
   NetEq::Config config;
   config.sample_rate_hz = kSampRateHz;
   NetEq* neteq = NetEq::Create(config);
   // Register decoder in |neteq|.
-  if (neteq->RegisterPayloadType(kDecoderType, kPayloadType) != 0)
+  if (neteq->RegisterPayloadType(kDecoderType, kDecoderName, kPayloadType) != 0)
     return -1;
 
   // Set up AudioLoop object.
   AudioLoop audio_loop;
   const size_t kMaxLoopLengthSamples = kSampRateHz * 10;  // 10 second loop.
   const size_t kInputBlockSizeSamples = 60 * kSampRateHz / 1000;  // 60 ms.
   if (!audio_loop.Init(kInputFileName, kMaxLoopLengthSamples,
                        kInputBlockSizeSamples))
     return -1;
 
@@ skipping 69 matching lines @@
       drift_flipped = true;
     }
   }
   int64_t end_time_ms = clock->TimeInMilliseconds();
   delete neteq;
   return end_time_ms - start_time_ms;
 }
 
 }  // namespace test
 }  // namespace webrtc