| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| D | sync_buffer_unittest.cc | 20 static const size_t kChannels = 2; in TEST() local
21 SyncBuffer sync_buffer(kChannels, kLen); in TEST()
22 EXPECT_EQ(kChannels, sync_buffer.Channels()); in TEST()
27 for (size_t channel = 0; channel < kChannels; ++channel) { in TEST()
37 static const size_t kChannels = 2; in TEST() local
38 SyncBuffer sync_buffer(kChannels, kLen); in TEST()
53 static const size_t kChannels = 2; in TEST() local
54 SyncBuffer sync_buffer(kChannels, kLen); in TEST()
56 AudioMultiVector new_data(kChannels, kNewLen); in TEST()
58 for (size_t channel = 0; channel < kChannels; ++channel) { in TEST()
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| D | dsp_helper_unittest.cc | 50 static const int kChannels = 5; in TEST() local
51 AudioMultiVector input(kChannels, kLen * 3); in TEST()
54 for (int channel = 0; channel < kChannels; ++channel) { in TEST()
72 for (int channel = 0; channel < kChannels; ++channel) { in TEST()
78 for (int channel = 0; channel < kChannels; ++channel) { in TEST()
84 for (int channel = 0; channel < kChannels; ++channel) { in TEST()
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| D | neteq_impl_unittest.cc | 802 static const size_t kChannels = 2; in TEST_F() local
812 int16_t dummy_output[kPayloadLengthSamples * kChannels] = {0}; in TEST_F()
825 size_t Channels() const override { return kChannels; } in TEST_F()
846 kPayloadLengthSamples * kChannels), in TEST_F()
849 kPayloadLengthSamples * kChannels)))); in TEST_F()
874 const size_t kMaxOutputSize = 10 * kSampleRateHz / 1000 * kChannels; in TEST_F()
884 EXPECT_EQ(kMaxOutputSize, samples_per_channel * kChannels); in TEST_F()
885 EXPECT_EQ(kChannels, num_channels); in TEST_F()
890 EXPECT_EQ(kMaxOutputSize, samples_per_channel * kChannels); in TEST_F()
891 EXPECT_EQ(kChannels, num_channels); in TEST_F()
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| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| D | input_audio_file_unittest.cc | 22 static const size_t kChannels = 2; in TEST() local
27 int16_t output[kSamples * kChannels]; in TEST()
28 InputAudioFile::DuplicateInterleaved(input, kSamples, kChannels, output); in TEST()
33 for (size_t j = 0; j < kChannels; ++j) { in TEST()
41 static const size_t kChannels = 5; in TEST() local
42 int16_t input[kSamples * kChannels]; in TEST()
46 InputAudioFile::DuplicateInterleaved(input, kSamples, kChannels, input); in TEST()
51 for (size_t j = 0; j < kChannels; ++j) { in TEST()
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| /external/webrtc/webrtc/common_audio/ |
| D | lapped_transform_unittest.cc | 93 const size_t kChannels = 3; in TEST() local
103 LappedTransform trans(kChannels, kChannels, kChunkLength, window, in TEST()
105 float in_buffer[kChannels][kChunkLength]; in TEST()
106 float* in_chunk[kChannels]; in TEST()
107 float out_buffer[kChannels][kChunkLength]; in TEST()
108 float* out_chunk[kChannels]; in TEST()
116 SetFloatArray(2.0f, kChannels, kChunkLength, in_chunk); in TEST()
117 SetFloatArray(-1.0f, kChannels, kChunkLength, out_chunk); in TEST()
121 for (size_t i = 0; i < kChannels; ++i) { in TEST()
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| D | audio_converter_unittest.cc | 146 const size_t kChannels[] = {1, 2}; in TEST() local
149 for (size_t src_channel = 0; src_channel < arraysize(kChannels); in TEST()
151 for (size_t dst_channel = 0; dst_channel < arraysize(kChannels); in TEST()
153 RunAudioConverterTest(kChannels[src_channel], kSampleRates[src_rate], in TEST()
154 kChannels[dst_channel], kSampleRates[dst_rate]); in TEST()
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| /external/webrtc/webrtc/common_audio/resampler/ |
| D | resampler_unittest.cc | 93 const int kChannels = 1; in TEST_F() local
103 EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kChannels)); in TEST_F()
108 EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kChannels)); in TEST_F()
115 const int kChannels = 2; in TEST_F() local
123 size_t in_length = static_cast<size_t>(kChannels * kRates[i] / 100); in TEST_F()
126 kChannels)); in TEST_F()
129 EXPECT_EQ(static_cast<size_t>(kChannels * kRates[j] / 100), out_length); in TEST_F()
132 kChannels)); in TEST_F()
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| /external/webrtc/webrtc/modules/audio_processing/ |
| D | splitting_filter_unittest.cc | 36 static const int kChannels = 1; in TEST() local
42 SplittingFilter splitting_filter(kChannels, in TEST()
45 IFChannelBuffer in_data(kSamplesPer48kHzChannel, kChannels, kNumBands); in TEST()
46 IFChannelBuffer bands(kSamplesPer48kHzChannel, kChannels, kNumBands); in TEST()
47 IFChannelBuffer out_data(kSamplesPer48kHzChannel, kChannels, kNumBands); in TEST()
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| /external/webrtc/webrtc/voice_engine/ |
| D | utility_unittest.cc | 210 const int kChannels[] = {1, 2}; in TEST_F() local
211 const int kChannelsSize = sizeof(kChannels) / sizeof(*kChannels); in TEST_F()
216 RunResampleTest(kChannels[src_channel], kSampleRates[src_rate], in TEST_F()
217 kChannels[dst_channel], kSampleRates[dst_rate]); in TEST_F()
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| /external/webrtc/webrtc/modules/audio_device/android/ |
| D | audio_manager_unittest.cc | 134 const size_t kChannels = 1; in TEST_F() local
139 AudioParameters params(kSampleRate, kChannels, kFramesPerBuffer); in TEST_F()
142 EXPECT_EQ(kChannels, params.channels()); in TEST_F()
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| /external/webrtc/webrtc/modules/audio_processing/test/ |
| D | audio_processing_unittest.cc | 56 const google::protobuf::int32 kChannels[] = {1, 2}; variable
2038 for (size_t i = 0; i < arraysize(kChannels); i++) { in TEST_F()
2039 for (size_t j = 0; j < arraysize(kChannels); j++) { in TEST_F()
2042 test->set_num_reverse_channels(kChannels[i]); in TEST_F()
2043 test->set_num_input_channels(kChannels[j]); in TEST_F()
2044 test->set_num_output_channels(kChannels[j]); in TEST_F()
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