Don't link with audio codecs that we don't use

webrtc/modules/audio_coding/neteq/neteq_unittest.cc

 /*
  *  Copyright (c) 2011 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 289 matching lines @@
 
 void NetEqDecodingTest::TearDown() {
   delete neteq_;
 }
 
 void NetEqDecodingTest::LoadDecoders() {
   // Load PCMu.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMu, 0));
   // Load PCMa.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCMa, 8));
-#ifndef WEBRTC_ANDROID
+#ifdef WEBRTC_CODEC_ILBC
   // Load iLBC.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderILBC, 102));
-#endif  // WEBRTC_ANDROID
+#endif
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
   // Load iSAC.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISAC, 103));
-#ifndef WEBRTC_ANDROID
+#endif
+#ifdef WEBRTC_CODEC_ISAC
   // Load iSAC SWB.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACswb, 104));
   // Load iSAC FB.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderISACfb, 105));
-#endif  // WEBRTC_ANDROID
+#endif
   // Load PCM16B nb.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16B, 93));
   // Load PCM16B wb.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bwb, 94));
   // Load PCM16B swb32.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderPCM16Bswb32kHz, 95));
   // Load CNG 8 kHz.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGnb, 13));
   // Load CNG 16 kHz.
   ASSERT_EQ(0, neteq_->RegisterPayloadType(kDecoderCNGwb, 98));
@@ skipping 102 matching lines @@
                                     size_t* payload_len) {
   rtp_info->header.sequenceNumber = frame_index;
   rtp_info->header.timestamp = timestamp;
   rtp_info->header.ssrc = 0x1234;  // Just an arbitrary SSRC.
   rtp_info->header.payloadType = 98;  // WB CNG.
   rtp_info->header.markerBit = 0;
   payload[0] = 64;  // Noise level -64 dBov, quite arbitrarily chosen.
   *payload_len = 1;  // Only noise level, no spectral parameters.
 }
 
+#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISAC)) && \
+    defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722)
+#define IF_ALL_CODECS(x) x
+#else
+#define IF_ALL_CODECS(x) DISABLED_##x
+#endif
+
 TEST_F(NetEqDecodingTest,
-       DISABLED_ON_IOS(DISABLED_ON_ANDROID(TestBitExactness))) {
+       DISABLED_ON_IOS(DISABLED_ON_ANDROID(IF_ALL_CODECS(TestBitExactness)))) {
   const std::string input_rtp_file = webrtc::test::ProjectRootPath() +
       "resources/audio_coding/neteq_universal_new.rtp";
   // Note that neteq4_universal_ref.pcm and neteq4_universal_ref_win_32.pcm
   // are identical. The latter could have been removed, but if clients still
   // have a copy of the file, the test will fail.
   const std::string input_ref_file =
       webrtc::test::ResourcePath("audio_coding/neteq4_universal_ref", "pcm");
 #if defined(_MSC_VER) && (_MSC_VER >= 1700)
   // For Visual Studio 2012 and later, we will have to use the generic reference
   // file, rather than the windows-specific one.
@@ skipping 357 matching lines @@
   const size_t kPayloadBytes = 100;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
   rtp_info.header.payloadType = 1;  // Not registered as a decoder.
   EXPECT_EQ(NetEq::kFail,
             neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
   EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
 }
 
-TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(DecoderError)) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+#define IF_ISAC(x) x
+#else
+#define IF_ISAC(x) DISABLED_##x
+#endif
+
+TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(IF_ISAC(DecoderError))) {
   const size_t kPayloadBytes = 100;
   uint8_t payload[kPayloadBytes] = {0};
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
   rtp_info.header.payloadType = 103;  // iSAC, but the payload is invalid.
   EXPECT_EQ(0, neteq_->InsertPacket(rtp_info, payload, kPayloadBytes, 0));
   NetEqOutputType type;
   // Set all of |out_data_| to 1, and verify that it was set to 0 by the call
   // to GetAudio.
   for (size_t i = 0; i < kMaxBlockSize; ++i) {
@@ skipping 210 matching lines @@
   CheckBgn(16000);
   CheckBgn(32000);
 }
 
 TEST_F(NetEqBgnTestFade, RunTest) {
   CheckBgn(8000);
   CheckBgn(16000);
   CheckBgn(32000);
 }
 
-TEST_F(NetEqDecodingTest, SyncPacketInsert) {
+TEST_F(NetEqDecodingTest, IF_ISAC(SyncPacketInsert)) {
   WebRtcRTPHeader rtp_info;
   uint32_t receive_timestamp = 0;
   // For the readability use the following payloads instead of the defaults of
   // this test.
   uint8_t kPcm16WbPayloadType = 1;
   uint8_t kCngNbPayloadType = 2;
   uint8_t kCngWbPayloadType = 3;
   uint8_t kCngSwb32PayloadType = 4;
   uint8_t kCngSwb48PayloadType = 5;
   uint8_t kAvtPayloadType = 6;
@@ skipping 447 matching lines @@
     // Pull audio once.
     ASSERT_EQ(0, neteq_->GetAudio(kMaxBlockSize, out_data_, &out_len,
                                   &num_channels, &type));
     ASSERT_EQ(kBlockSize16kHz, out_len);
   }
   // Verify speech output.
   EXPECT_EQ(kOutputNormal, type);
 }
 
 }  // namespace webrtc