Fixing a unittest to allow new Opus version.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc

 /*
  *  Copyright (c) 2015 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 256 matching lines @@
 
 TEST(AudioEncoderOpusTest, SetReceiverFrameLengthRange) {
   auto states = CreateCodec(2);
   // Before calling to |SetReceiverFrameLengthRange|,
   // |supported_frame_lengths_ms| should contain only the frame length being
   // used.
   using ::testing::ElementsAre;
   EXPECT_THAT(states.encoder->supported_frame_lengths_ms(),
               ElementsAre(states.encoder->next_frame_length_ms()));
   states.encoder->SetReceiverFrameLengthRange(0, 12345);
-  EXPECT_THAT(states.encoder->supported_frame_lengths_ms(),
-              ElementsAre(20, 60));
   states.encoder->SetReceiverFrameLengthRange(21, 60);
   EXPECT_THAT(states.encoder->supported_frame_lengths_ms(), ElementsAre(60));
   states.encoder->SetReceiverFrameLengthRange(20, 59);
   EXPECT_THAT(states.encoder->supported_frame_lengths_ms(), ElementsAre(20));
 }
 
 TEST(AudioEncoderOpusTest,
      InvokeAudioNetworkAdaptorOnReceivedUplinkPacketLossFraction) {
   auto states = CreateCodec(2);
   states.encoder->EnableAudioNetworkAdaptor("", nullptr, nullptr);
@@ skipping 250 matching lines @@
 
     // Should encode now.
     states.encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
                            &encoded);
     EXPECT_GT(encoded.size(), 0u);
     encoded.Clear();
   }
 }
 
 }  // namespace webrtc