Remove ACMCodecDB::Codec, and make the rest of ACMCodecDB private

webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc

 /*
  *  Copyright (c) 2013 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 70 matching lines @@
     rtp_header_.header.ssrc = 0x12345678;  // Arbitrary.
     rtp_header_.header.numCSRCs = 0;
     rtp_header_.header.payloadType = 0;
     rtp_header_.frameType = kAudioFrameSpeech;
     rtp_header_.type.Audio.isCNG = false;
   }
 
   void TearDown() override {}
 
   void InsertOnePacketOfSilence(int codec_id) {
-    CodecInst codec;
-    ACMCodecDB::Codec(codec_id, &codec);
+    CodecInst codec =
+        *RentACodec::CodecInstById(*RentACodec::CodecIdFromIndex(codec_id));
     if (timestamp_ == 0) {  // This is the first time inserting audio.
       ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
     } else {
       auto current_codec = acm_->SendCodec();
       ASSERT_TRUE(current_codec);
       if (!CodecsEqual(codec, *current_codec))
         ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
     }
     AudioFrame frame;
     // Frame setup according to the codec.
@@ skipping 256 matching lines @@
     }
     EXPECT_EQ(c.id, receiver_->last_audio_codec_id());
     EXPECT_EQ(0, receiver_->LastAudioCodec(&codec));
     EXPECT_TRUE(CodecsEqual(c.inst, codec));
   }
 }
 
 }  // namespace acm2
 
 }  // namespace webrtc