Let AudioCodingModule::SendCodec return Maybe<CodecInst>

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 75 matching lines @@
   }
 
   void TearDown() override {}
 
   void InsertOnePacketOfSilence(int codec_id) {
     CodecInst codec;
     ACMCodecDB::Codec(codec_id, &codec);
     if (timestamp_ == 0) {  // This is the first time inserting audio.
       ASSERT_EQ(0, acm_->RegisterSendCodec(codec));
     } else {
-      CodecInst current_codec;
-      ASSERT_EQ(0, acm_->SendCodec(&current_codec));
-      if (!CodecsEqual(codec, current_codec))
+      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.
     frame.sample_rate_hz_ = codec.plfreq;
     frame.samples_per_channel_ = codec.plfreq / 100;  // 10 ms.
     frame.num_channels_ = codec.channels;
     memset(frame.data_, 0, frame.samples_per_channel_ * frame.num_channels_ *
            sizeof(int16_t));
     packet_sent_ = false;
@@ skipping 250 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