RTPPayloadRegistry: Stop using the rate to keep track of receive codecs

webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc

 /*
  *  Copyright (c) 2012 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 57 matching lines @@
   }
 };
 
 class RTPCallback : public NullRtpFeedback {
  public:
   int32_t OnInitializeDecoder(const int8_t payloadType,
                               const char payloadName[RTP_PAYLOAD_NAME_SIZE],
                               const int frequency,
                               const size_t channels,
                               const uint32_t rate) override {
-    if (payloadType == kPcmuPayloadType) {
-      EXPECT_EQ(kTestRate, rate) <<
-          "The rate should be 64K for this payloadType";
-    }
+    EXPECT_EQ(0u, rate) << "The rate should be zero";
     return 0;
   }
 };
 
 }  // namespace
 
 class RtpRtcpAudioTest : public ::testing::Test {
  protected:
   RtpRtcpAudioTest()
       : fake_clock(123456), retransmission_rate_limiter_(&fake_clock, 1000) {
@@ skipping 189 matching lines @@
                                           nullptr, nullptr, nullptr));
 
     EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
     EXPECT_TRUE(rtp_receiver2_->Timestamp(&timestamp));
     EXPECT_EQ(test_timestamp + in_timestamp, timestamp);
     in_timestamp += 10;
   }
 }
 
 }  // namespace webrtc