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Diffstat (limited to 'third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc | 168 |
1 files changed, 168 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc new file mode 100644 index 0000000000..4a2b261a59 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc @@ -0,0 +1,168 @@ +/* + * Copyright (c) 2016 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. + */ + +#include "api/audio_codecs/builtin_audio_decoder_factory.h" + +#include <memory> + +#include "test/gtest.h" + +namespace webrtc { + +TEST(AudioDecoderFactoryTest, CreateUnknownDecoder) { + rtc::scoped_refptr<AudioDecoderFactory> adf = + CreateBuiltinAudioDecoderFactory(); + ASSERT_TRUE(adf); + EXPECT_FALSE( + adf->MakeAudioDecoder(SdpAudioFormat("rey", 8000, 1), absl::nullopt)); +} + +TEST(AudioDecoderFactoryTest, CreatePcmu) { + rtc::scoped_refptr<AudioDecoderFactory> adf = + CreateBuiltinAudioDecoderFactory(); + ASSERT_TRUE(adf); + // PCMu supports 8 kHz, and any number of channels. + EXPECT_FALSE( + adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 0), absl::nullopt)); + EXPECT_TRUE( + adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 1), absl::nullopt)); + EXPECT_TRUE( + adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 2), absl::nullopt)); + EXPECT_TRUE( + adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 3), absl::nullopt)); + EXPECT_FALSE( + adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 16000, 1), absl::nullopt)); +} + +TEST(AudioDecoderFactoryTest, CreatePcma) { + rtc::scoped_refptr<AudioDecoderFactory> adf = + CreateBuiltinAudioDecoderFactory(); + ASSERT_TRUE(adf); + // PCMa supports 8 kHz, and any number of channels. + EXPECT_FALSE( + adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 0), absl::nullopt)); + EXPECT_TRUE( + adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 1), absl::nullopt)); + EXPECT_TRUE( + adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 2), absl::nullopt)); + EXPECT_TRUE( + adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 3), absl::nullopt)); + EXPECT_FALSE( + adf->MakeAudioDecoder(SdpAudioFormat("pcma", 16000, 1), absl::nullopt)); +} + +TEST(AudioDecoderFactoryTest, CreateIlbc) { + rtc::scoped_refptr<AudioDecoderFactory> adf = + CreateBuiltinAudioDecoderFactory(); + ASSERT_TRUE(adf); + // iLBC supports 8 kHz, 1 channel. + EXPECT_FALSE( + adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 0), absl::nullopt)); +#ifdef WEBRTC_CODEC_ILBC + EXPECT_TRUE( + adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 1), absl::nullopt)); +#endif + EXPECT_FALSE( + adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 2), absl::nullopt)); + EXPECT_FALSE( + adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 16000, 1), absl::nullopt)); +} + +TEST(AudioDecoderFactoryTest, CreateL16) { + rtc::scoped_refptr<AudioDecoderFactory> adf = + CreateBuiltinAudioDecoderFactory(); + ASSERT_TRUE(adf); + // L16 supports any clock rate and any number of channels up to 24. + const int clockrates[] = {8000, 16000, 32000, 48000}; + const int num_channels[] = {1, 2, 3, 24}; + for (int clockrate : clockrates) { + EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, 0), + absl::nullopt)); + for (int channels : num_channels) { + EXPECT_TRUE(adf->MakeAudioDecoder( + SdpAudioFormat("l16", clockrate, channels), absl::nullopt)); + } + } +} + +// Tests that using more channels than the maximum does not work +TEST(AudioDecoderFactoryTest, MaxNrOfChannels) { + rtc::scoped_refptr<AudioDecoderFactory> adf = + CreateBuiltinAudioDecoderFactory(); + std::vector<std::string> codecs = { +#ifdef WEBRTC_CODEC_OPUS + "opus", +#endif +#ifdef WEBRTC_CODEC_ILBC + "ilbc", +#endif + "pcmu", + "pcma", + "l16", + "G722", + "G711", + }; + + for (auto codec : codecs) { + EXPECT_FALSE(adf->MakeAudioDecoder( + SdpAudioFormat(codec, 32000, AudioDecoder::kMaxNumberOfChannels + 1), + absl::nullopt)); + } +} + +TEST(AudioDecoderFactoryTest, CreateG722) { + rtc::scoped_refptr<AudioDecoderFactory> adf = + CreateBuiltinAudioDecoderFactory(); + ASSERT_TRUE(adf); + // g722 supports 8 kHz, 1-2 channels. + EXPECT_FALSE( + adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 0), absl::nullopt)); + EXPECT_TRUE( + adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), absl::nullopt)); + EXPECT_TRUE( + adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 2), absl::nullopt)); + EXPECT_FALSE( + adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 3), absl::nullopt)); + EXPECT_FALSE( + adf->MakeAudioDecoder(SdpAudioFormat("g722", 16000, 1), absl::nullopt)); + EXPECT_FALSE( + adf->MakeAudioDecoder(SdpAudioFormat("g722", 32000, 1), absl::nullopt)); + + // g722 actually uses a 16 kHz sample rate instead of the nominal 8 kHz. + std::unique_ptr<AudioDecoder> dec = + adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), absl::nullopt); + EXPECT_EQ(16000, dec->SampleRateHz()); +} + +TEST(AudioDecoderFactoryTest, CreateOpus) { + rtc::scoped_refptr<AudioDecoderFactory> adf = + CreateBuiltinAudioDecoderFactory(); + ASSERT_TRUE(adf); + // Opus supports 48 kHz, 2 channels, and wants a "stereo" parameter whose + // value is either "0" or "1". + for (int hz : {8000, 16000, 32000, 48000}) { + for (int channels : {0, 1, 2, 3}) { + for (std::string stereo : {"XX", "0", "1", "2"}) { + SdpAudioFormat::Parameters params; + if (stereo != "XX") { + params["stereo"] = stereo; + } + const bool good = (hz == 48000 && channels == 2 && + (stereo == "XX" || stereo == "0" || stereo == "1")); + EXPECT_EQ(good, + static_cast<bool>(adf->MakeAudioDecoder( + SdpAudioFormat("opus", hz, channels, std::move(params)), + absl::nullopt))); + } + } + } +} + +} // namespace webrtc |