Let AudioCodingModule::SendCodec return Maybe<CodecInst>

webrtc/modules/audio_coding/main/test/iSACTest.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 29 matching lines @@
   isacConfig.initFrameSizeInMsec = 0;
   isacConfig.enforceFrameSize = false;
   return;
 }
 
 int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
                      int testMode) {
 
   if ((isacConfig.currentRateBitPerSec != 0)
       || (isacConfig.currentFrameSizeMsec != 0)) {
-    CodecInst sendCodec;
-    EXPECT_EQ(0, acm->SendCodec(&sendCodec));
+    auto sendCodec = acm->SendCodec();
+    EXPECT_TRUE(sendCodec);
     if (isacConfig.currentRateBitPerSec < 0) {
       // Register iSAC in adaptive (channel-dependent) mode.
-      sendCodec.rate = -1;
-      EXPECT_EQ(0, acm->RegisterSendCodec(sendCodec));
+      sendCodec->rate = -1;
+      EXPECT_EQ(0, acm->RegisterSendCodec(*sendCodec));
     } else {
       if (isacConfig.currentRateBitPerSec != 0) {
-        sendCodec.rate = isacConfig.currentRateBitPerSec;
+        sendCodec->rate = isacConfig.currentRateBitPerSec;
       }
       if (isacConfig.currentFrameSizeMsec != 0) {
-        sendCodec.pacsize = isacConfig.currentFrameSizeMsec
-            * (sendCodec.plfreq / 1000);
+        sendCodec->pacsize = isacConfig.currentFrameSizeMsec
+            * (sendCodec->plfreq / 1000);
       }
-      EXPECT_EQ(0, acm->RegisterSendCodec(sendCodec));
+      EXPECT_EQ(0, acm->RegisterSendCodec(*sendCodec));
     }
   }
 
   return 0;
 }
 
 ISACTest::ISACTest(int testMode)
     : _acmA(AudioCodingModule::Create(1)),
       _acmB(AudioCodingModule::Create(2)),
       _testMode(testMode) {}
@@ skipping 156 matching lines @@
   bool adaptiveMode = false;
   if ((swbISACConfig.currentRateBitPerSec == -1)
       || (wbISACConfig.currentRateBitPerSec == -1)) {
     adaptiveMode = true;
   }
   _myTimer.Reset();
   _channel_A2B->ResetStats();
   _channel_B2A->ResetStats();
 
   char currentTime[500];
-  CodecInst sendCodec;
   EventTimerWrapper* myEvent = EventTimerWrapper::Create();
   EXPECT_TRUE(myEvent->StartTimer(true, 10));
   while (!(_inFileA.EndOfFile() || _inFileA.Rewinded())) {
     Run10ms();
     _myTimer.Tick10ms();
     _myTimer.CurrentTimeHMS(currentTime);
 
     if ((adaptiveMode) && (_testMode != 0)) {
       myEvent->Wait(5000);
-      EXPECT_EQ(0, _acmA->SendCodec(&sendCodec));
-      EXPECT_EQ(0, _acmB->SendCodec(&sendCodec));
+      EXPECT_TRUE(_acmA->SendCodec());
+      EXPECT_TRUE(_acmB->SendCodec());
     }
   }
 
   if (_testMode != 0) {
     printf("\n\nSide A statistics\n\n");
     _channel_A2B->PrintStats(_paramISAC32kHz);
 
     printf("\n\nSide B statistics\n\n");
     _channel_B2A->PrintStats(_paramISAC16kHz);
   }
@@ skipping 68 matching lines @@
       numSendCodecChanged++;
     }
   }
   _outFileA.Close();
   _outFileB.Close();
   _inFileA.Close();
   _inFileB.Close();
 }
 
 }  // namespace webrtc