NetEq: Add codec name and RTP timestamp rate to DecoderInfo

webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_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 847 matching lines @@
     return others;
 #endif
   }
 
  protected:
   struct ExternalDecoder {
     int rtp_payload_type;
     AudioDecoder* external_decoder;
     int sample_rate_hz;
     int num_channels;
+    std::string name;
   };
 
   void Run(int output_freq_hz,
            const std::string& checksum_ref,
            const std::vector<ExternalDecoder>& external_decoders) {
     const std::string input_file_name =
         webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
     rtc::scoped_ptr<test::RtpFileSource> packet_source(
         test::RtpFileSource::Create(input_file_name));
 #ifdef WEBRTC_ANDROID
@@ skipping 15 matching lines @@
 
     test::AcmReceiveTestOldApi test(
         packet_source.get(),
         &output,
         output_freq_hz,
         test::AcmReceiveTestOldApi::kArbitraryChannels);
     ASSERT_NO_FATAL_FAILURE(test.RegisterNetEqTestCodecs());
     for (const auto& ed : external_decoders) {
       ASSERT_EQ(0, test.RegisterExternalReceiveCodec(
                        ed.rtp_payload_type, ed.external_decoder,
-                       ed.sample_rate_hz, ed.num_channels));
+                       ed.sample_rate_hz, ed.num_channels, ed.name));
     }
     test.Run();
 
     std::string checksum_string = checksum.Finish();
     EXPECT_EQ(checksum_ref, checksum_string);
   }
 };
 
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISAC)) && \
     defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
@@ skipping 79 matching lines @@
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::HasDecodePlc));
   EXPECT_CALL(mock_decoder, PacketDuration(_, _))
       .Times(AtLeast(1))
       .WillRepeatedly(Invoke(&decoder, &AudioDecoderPcmU::PacketDuration));
   ExternalDecoder ed;
   ed.rtp_payload_type = 0;
   ed.external_decoder = &mock_decoder;
   ed.sample_rate_hz = 8000;
   ed.num_channels = 1;
+  ed.name = "MockPCMU";
   std::vector<ExternalDecoder> external_decoders;
   external_decoders.push_back(ed);
 
   Run(48000, PlatformChecksum("aa7c232f63a67b2a72703593bdd172e0",
                               "0155665e93067c4e89256b944dd11999",
                               "4ee2730fa1daae755e8a8fd3abd779ec"),
       external_decoders);
 
   EXPECT_CALL(mock_decoder, Die());
 }
@@ skipping 749 matching lines @@
   Run(16000, 8000, 1000);
 }
 
 TEST_F(AcmSwitchingOutputFrequencyOldApi, Toggle8KhzTo16Khz) {
   Run(8000, 16000, 1000);
 }
 
 #endif
 
 }  // namespace webrtc