Remove DISABLED_ON_ macros.

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 12 matching lines @@
 #include <string>
 #include <vector>
 
 #include "gflags/gflags.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/neteq/tools/audio_loop.h"
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
 #include "webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h"
 #include "webrtc/test/testsupport/fileutils.h"
-#include "webrtc/test/testsupport/gtest_disable.h"
 #include "webrtc/typedefs.h"
 
 #ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
 #else
 #include "webrtc/audio_coding/neteq/neteq_unittest.pb.h"
 #endif
 #endif
 
@@ skipping 880 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, 0));
   EXPECT_EQ(NetEq::kUnknownRtpPayloadType, neteq_->LastError());
 }
 
 #if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
-#define IF_ISAC(x) x
+#if defined(WEBRTC_ANDROID)
+TEST_F(NetEqDecodingTest, DISABLED_DecoderError) {
 #else
-#define IF_ISAC(x) DISABLED_##x
+TEST_F(NetEqDecodingTest, DecoderError) {
 #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, 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 17 matching lines @@
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
     EXPECT_EQ(0, out_data_[i]);
   }
   for (size_t i = kExpectedOutputLength; i < kMaxBlockSize; ++i) {
     std::ostringstream ss;
     ss << "i = " << i;
     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
     EXPECT_EQ(1, out_data_[i]);
   }
 }
+#endif
 
 TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
   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) {
     out_data_[i] = 1;
   }
   int num_channels;
   size_t samples_per_channel;
@@ skipping 177 matching lines @@
   CheckBgn(16000);
   CheckBgn(32000);
 }
 
 TEST_F(NetEqBgnTestFade, RunTest) {
   CheckBgn(8000);
   CheckBgn(16000);
   CheckBgn(32000);
 }
 
-TEST_F(NetEqDecodingTest, IF_ISAC(SyncPacketInsert)) {
+#if defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)
+TEST_F(NetEqDecodingTest, 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 58 matching lines @@
   EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
 
   // Change of SSRC is not allowed with a sync packet.
   rtp_info.header.payloadType = kPcm16WbPayloadType;
   ++rtp_info.header.ssrc;
   EXPECT_EQ(-1, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
 
   --rtp_info.header.ssrc;
   EXPECT_EQ(0, neteq_->InsertSyncPacket(rtp_info, receive_timestamp));
 }
+#endif
 
 // First insert several noise like packets, then sync-packets. Decoding all
 // packets should not produce error, statistics should not show any packet loss
 // and sync-packets should decode to zero.
 // TODO(turajs) we will have a better test if we have a referece NetEq, and
 // when Sync packets are inserted in "test" NetEq we insert all-zero payload
 // in reference NetEq and compare the output of those two.
 TEST_F(NetEqDecodingTest, SyncPacketDecode) {
   WebRtcRTPHeader rtp_info;
   PopulateRtpInfo(0, 0, &rtp_info);
@@ skipping 374 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