Searched refs:kFrameSizeSamples (Results 1 – 6 of 6) sorted by relevance
| /external/webrtc/webrtc/modules/audio_coding/test/ |
| D | target_delay_unittest.cc | 46 int16_t audio[kFrameSizeSamples]; in SetUp()
48 for (size_t n = 0; n < kFrameSizeSamples; ++n) in SetUp()
50 WebRtcPcm16b_Encode(audio, kFrameSizeSamples, payload_); in SetUp()
135 static const size_t kFrameSizeSamples = 320; // 20 ms @ 16 kHz. member in webrtc::TargetDelayTest
143 rtp_info_.header.timestamp += kFrameSizeSamples; in Push()
145 ASSERT_EQ(0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2, in Push()
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| /external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
| D | neteq_ilbc_quality_test.cc | 64 const size_t kFrameSizeSamples = 80; // Samples per 10 ms. in EncodeBlock() local
71 in_data + encoded_samples, kFrameSizeSamples), in EncodeBlock()
73 encoded_samples += kFrameSizeSamples; in EncodeBlock()
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| D | neteq_pcmu_quality_test.cc | 64 const size_t kFrameSizeSamples = 80; // Samples per 10 ms. in EncodeBlock() local
71 in_data + encoded_samples, kFrameSizeSamples), in EncodeBlock()
73 encoded_samples += kFrameSizeSamples; in EncodeBlock()
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| D | audio_classifier_test.cc | 37 const int kFrameSizeSamples = 960; in main() local
50 const int data_size = channels * kFrameSizeSamples; in main()
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| /external/webrtc/webrtc/modules/audio_device/ |
| D | fine_audio_buffer_unittest.cc | 136 const int kFrameSizeSamples = kSamplesPer10Ms - 50; in TEST() local
137 RunFineBufferTest(kSampleRate, kFrameSizeSamples); in TEST()
143 const int kFrameSizeSamples = kSamplesPer10Ms + 50; in TEST() local
144 RunFineBufferTest(kSampleRate, kFrameSizeSamples); in TEST()
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| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| D | audio_coding_module_unittest_oldapi.cc | 55 const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms; variable
56 const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
158 rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)), in AudioCodingModuleTestOldApi()
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