Don't select audio codecs depending on GN vars `build_with_{chromium|mozilla}`

BUG=webrtc:8343

Change-Id: I5943006a4da17f72eb88eae9d7ea57574d54f680
Reviewed-on: https://webrtc-review.googlesource.com/9401
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20540}
diff --git a/modules/audio_coding/test/TestAllCodecs.cc b/modules/audio_coding/test/TestAllCodecs.cc
index ff28a28..f7f220f 100644
--- a/modules/audio_coding/test/TestAllCodecs.cc
+++ b/modules/audio_coding/test/TestAllCodecs.cc
@@ -151,7 +151,6 @@
 
   // All codecs are tested for all allowed sampling frequencies, rates and
   // packet sizes.
-#ifdef WEBRTC_CODEC_G722
   if (test_mode_ != 0) {
     printf("===============================================================\n");
   }
@@ -171,7 +170,6 @@
   RegisterSendCodec('A', codec_g722, 16000, 64000, 960, 0);
   Run(channel_a_to_b_);
   outfile_b_.Close();
-#endif
 #ifdef WEBRTC_CODEC_ILBC
   if (test_mode_ != 0) {
     printf("===============================================================\n");
@@ -324,9 +322,6 @@
 
     /* Print out all codecs that were not tested in the run */
     printf("The following codecs was not included in the test:\n");
-#ifndef WEBRTC_CODEC_G722
-    printf("   G.722\n");
-#endif
 #ifndef WEBRTC_CODEC_ILBC
     printf("   iLBC\n");
 #endif
diff --git a/modules/audio_coding/test/TestRedFec.cc b/modules/audio_coding/test/TestRedFec.cc
index 58561c6..034c595 100644
--- a/modules/audio_coding/test/TestRedFec.cc
+++ b/modules/audio_coding/test/TestRedFec.cc
@@ -38,13 +38,9 @@
   const char kNamePCMU[] = "PCMU";
   const char kNameCN[] = "CN";
   const char kNameRED[] = "RED";
-
-  // These three are only used by code #ifdeffed on WEBRTC_CODEC_G722.
-#ifdef WEBRTC_CODEC_G722
   const char kNameISAC[] = "ISAC";
   const char kNameG722[] = "G722";
   const char kNameOPUS[] = "opus";
-#endif
 }
 
 TestRedFec::TestRedFec()
@@ -104,11 +100,6 @@
   Run();
   _outFileB.Close();
 
-#ifndef WEBRTC_CODEC_G722
-  EXPECT_TRUE(false);
-  printf("G722 needs to be activated to run this test\n");
-  return;
-#else
   EXPECT_EQ(0, RegisterSendCodec('A', kNameG722, 16000));
   EXPECT_EQ(0, RegisterSendCodec('A', kNameCN, 16000));
 
@@ -412,8 +403,6 @@
   EXPECT_FALSE(_acmA->REDStatus());
   EXPECT_EQ(0, _acmA->SetCodecFEC(false));
   EXPECT_FALSE(_acmA->CodecFEC());
-
-#endif  // defined(WEBRTC_CODEC_G722)
 }
 
 int32_t TestRedFec::SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode) {
diff --git a/modules/audio_coding/test/TestStereo.cc b/modules/audio_coding/test/TestStereo.cc
index eca81f8..ba86719 100644
--- a/modules/audio_coding/test/TestStereo.cc
+++ b/modules/audio_coding/test/TestStereo.cc
@@ -114,13 +114,11 @@
       test_cntr_(0),
       pack_size_samp_(0),
       pack_size_bytes_(0),
-      counter_(0)
-#ifdef WEBRTC_CODEC_G722
-      , g722_pltype_(0)
-#endif
-      , l16_8khz_pltype_(-1)
-      , l16_16khz_pltype_(-1)
-      , l16_32khz_pltype_(-1)
+      counter_(0),
+      g722_pltype_(0),
+      l16_8khz_pltype_(-1),
+      l16_16khz_pltype_(-1),
+      l16_32khz_pltype_(-1)
 #ifdef PCMA_AND_PCMU
       , pcma_pltype_(-1)
       , pcmu_pltype_(-1)
@@ -128,7 +126,7 @@
 #ifdef WEBRTC_CODEC_OPUS
       , opus_pltype_(-1)
 #endif
-      {
+{
   // test_mode = 0 for silent test (auto test)
   test_mode_ = test_mode;
 }
@@ -217,7 +215,6 @@
 
   // All codecs are tested for all allowed sampling frequencies, rates and
   // packet sizes.
-#ifdef WEBRTC_CODEC_G722
   if (test_mode_ != 0) {
     printf("===========================================================\n");
     printf("Test number: %d\n", test_cntr_ + 1);
@@ -246,7 +243,7 @@
       g722_pltype_);
   Run(channel_a2b_, audio_channels, codec_channels);
   out_file_.Close();
-#endif
+
   if (test_mode_ != 0) {
     printf("===========================================================\n");
     printf("Test number: %d\n", test_cntr_ + 1);
@@ -419,7 +416,6 @@
   audio_channels = 1;
   codec_channels = 2;
 
-#ifdef WEBRTC_CODEC_G722
   if (test_mode_ != 0) {
     printf("===============================================================\n");
     printf("Test number: %d\n", test_cntr_ + 1);
@@ -432,7 +428,7 @@
       g722_pltype_);
   Run(channel_a2b_, audio_channels, codec_channels);
   out_file_.Close();
-#endif
+
   if (test_mode_ != 0) {
     printf("===============================================================\n");
     printf("Test number: %d\n", test_cntr_ + 1);
@@ -512,7 +508,6 @@
   codec_channels = 1;
   channel_a2b_->set_codec_mode(kMono);
 
-#ifdef WEBRTC_CODEC_G722
   // Run stereo audio and mono codec.
   if (test_mode_ != 0) {
     printf("===============================================================\n");
@@ -533,7 +528,7 @@
   EXPECT_EQ(0, acm_a_->SetVAD(false, false, VADNormal));
   Run(channel_a2b_, audio_channels, codec_channels);
   out_file_.Close();
-#endif
+
   if (test_mode_ != 0) {
     printf("===============================================================\n");
     printf("Test number: %d\n", test_cntr_ + 1);
@@ -659,9 +654,7 @@
   // Print out which codecs were tested, and which were not, in the run.
   if (test_mode_ != 0) {
     printf("\nThe following codecs was INCLUDED in the test:\n");
-#ifdef WEBRTC_CODEC_G722
     printf("   G.722\n");
-#endif
     printf("   PCM16\n");
     printf("   G.711\n");
 #ifdef WEBRTC_CODEC_OPUS
diff --git a/modules/audio_coding/test/TestStereo.h b/modules/audio_coding/test/TestStereo.h
index a27d8d7..a454f25 100644
--- a/modules/audio_coding/test/TestStereo.h
+++ b/modules/audio_coding/test/TestStereo.h
@@ -98,9 +98,7 @@
   char* send_codec_name_;
 
   // Payload types for stereo codecs and CNG
-#ifdef WEBRTC_CODEC_G722
   int g722_pltype_;
-#endif
   int l16_8khz_pltype_;
   int l16_16khz_pltype_;
   int l16_32khz_pltype_;
diff --git a/modules/audio_coding/test/Tester.cc b/modules/audio_coding/test/Tester.cc
index 73625f1..9cd774d 100644
--- a/modules/audio_coding/test/Tester.cc
+++ b/modules/audio_coding/test/Tester.cc
@@ -63,7 +63,7 @@
 #endif
 
 #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
-    defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
+    defined(WEBRTC_CODEC_ILBC)
 #if defined(WEBRTC_ANDROID)
 TEST(AudioCodingModuleTest, DISABLED_TwoWayCommunication) {
 #else