Chromium Code Reviews| Index: webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
| diff --git a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
| index 04cef8aa99e7f1de8f2fbce8a7f3004aeaada010..09df04691f5560c08b138ea4b11193b4fa68110d 100644 |
| --- a/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
| +++ b/webrtc/modules/audio_coding/neteq/neteq_unittest.cc |
| @@ -408,6 +408,10 @@ void NetEqDecodingTest::Process(size_t* out_len) { |
| if (packet_->payload_length_bytes() > 0) { |
| WebRtcRTPHeader rtp_header; |
| packet_->ConvertHeader(&rtp_header); |
| +#ifdef WEBRTC_ANDROID |
| + // Payload type 104 is not supported on android. |
| + if (rtp_header.header.payloadType != 104) |
| +#endif |
| ASSERT_EQ(0, neteq_->InsertPacket( |
| rtp_header, |
| rtc::ArrayView<const uint8_t>( |
| @@ -510,14 +514,14 @@ void NetEqDecodingTest::PopulateCng(int frame_index, |
| *payload_len = 1; // Only noise level, no spectral parameters. |
| } |
| -#if !defined(WEBRTC_IOS) && !defined(WEBRTC_ANDROID) && \ |
| - defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \ |
| - (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ |
| +#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \ |
| + (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \ |
| defined(WEBRTC_CODEC_ILBC) && defined(WEBRTC_CODEC_G722) |
| #define MAYBE_TestBitExactness TestBitExactness |
| #else |
| #define MAYBE_TestBitExactness DISABLED_TestBitExactness |
| #endif |
| + |
| TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) { |
| const std::string input_rtp_file = webrtc::test::ProjectRootPath() + |
| "resources/audio_coding/neteq_universal_new.rtp"; |
| @@ -902,7 +906,7 @@ TEST_F(NetEqDecodingTest, UnknownPayloadType) { |
| #define IF_ISAC(x) DISABLED_##x |
| #endif |
| -TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(IF_ISAC(DecoderError))) { |
| +TEST_F(NetEqDecodingTest, IF_ISAC(DecoderError)) { |
|
minyue-webrtc
2015/12/17 14:32:58
may you change this to MAYBE_?
#if defined(WEBRTC
ivoc
2015/12/17 15:06:28
Good idea, done.
|
| const size_t kPayloadBytes = 100; |
| uint8_t payload[kPayloadBytes] = {0}; |
| WebRtcRTPHeader rtp_info; |
| @@ -922,8 +926,12 @@ TEST_F(NetEqDecodingTest, DISABLED_ON_ANDROID(IF_ISAC(DecoderError))) { |
| &samples_per_channel, &num_channels, &type)); |
| // Verify that there is a decoder error to check. |
| EXPECT_EQ(NetEq::kDecoderErrorCode, neteq_->LastError()); |
| - // Code 6730 is an iSAC error code. |
| + |
| +#if defined(WEBRTC_CODEC_ISAC) |
| EXPECT_EQ(6730, neteq_->LastDecoderError()); |
|
minyue-webrtc
2015/12/17 14:32:58
would you use enum (see below) or const for these
ivoc
2015/12/17 15:06:28
Thanks for the suggestion, looks more clear.
|
| +#else |
|
minyue-webrtc
2015/12/17 14:32:58
maybe use
#elif defined(WEBRTC_CODEC_ISACFX)
ivoc
2015/12/17 15:06:27
Done.
|
| + EXPECT_EQ(6640, neteq_->LastDecoderError()); |
| +#endif |
| // Verify that the first 160 samples are set to 0, and that the remaining |
| // samples are left unmodified. |
| static const int kExpectedOutputLength = 160; // 10 ms at 16 kHz sample rate. |