Searched refs:kFrameSizeSamples (Results 1 – 8 of 8) sorted by relevance
| /external/webrtc/modules/audio_device/ |
| D | fine_audio_buffer_unittest.cc | 90 const int kFrameSizeSamples = frame_size_in_samples; in RunFineBufferTest() local
129 new int16_t[kChannels * kFrameSizeSamples]); in RunFineBufferTest()
131 new int16_t[kChannels * kFrameSizeSamples]); in RunFineBufferTest()
136 kChannels * kFrameSizeSamples), in RunFineBufferTest()
139 VerifyBuffer(out_buffer.get(), i, kChannels * kFrameSizeSamples)); in RunFineBufferTest()
140 UpdateInputBuffer(in_buffer.get(), i, kChannels * kFrameSizeSamples); in RunFineBufferTest()
143 kChannels * kFrameSizeSamples), in RunFineBufferTest()
149 const int kFrameSizeSamples = kSamplesPer10Ms - 50; in TEST() local
150 RunFineBufferTest(kFrameSizeSamples); in TEST()
154 const int kFrameSizeSamples = kSamplesPer10Ms + 50; in TEST() local
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| /external/webrtc/modules/audio_coding/test/ |
| D | target_delay_unittest.cc | 44 int16_t audio[kFrameSizeSamples]; in SetUp()
46 for (size_t n = 0; n < kFrameSizeSamples; ++n) in SetUp()
48 WebRtcPcm16b_Encode(audio, kFrameSizeSamples, payload_); in SetUp()
76 static const size_t kFrameSizeSamples = 320; // 20 ms @ 16 kHz. member in webrtc::TargetDelayTest
84 rtp_header_.timestamp += kFrameSizeSamples; in Push()
88 payload_, kFrameSizeSamples * 2))); in Push()
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| /external/webrtc/modules/audio_coding/neteq/test/ |
| D | neteq_pcmu_quality_test.cc | 57 const size_t kFrameSizeSamples = 80; // Samples per 10 ms. in EncodeBlock() local
64 in_data + encoded_samples, kFrameSizeSamples), in EncodeBlock()
66 encoded_samples += kFrameSizeSamples; in EncodeBlock()
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| D | neteq_ilbc_quality_test.cc | 58 const size_t kFrameSizeSamples = 80; // Samples per 10 ms. in EncodeBlock() local
65 in_data + encoded_samples, kFrameSizeSamples), in EncodeBlock()
67 encoded_samples += kFrameSizeSamples; in EncodeBlock()
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| D | neteq_pcm16b_quality_test.cc | 58 const size_t kFrameSizeSamples = 480; // Samples per 10 ms. in EncodeBlock() local
65 in_data + encoded_samples, kFrameSizeSamples), in EncodeBlock()
67 encoded_samples += kFrameSizeSamples; in EncodeBlock()
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| /external/webrtc/modules/audio_coding/acm2/ |
| D | acm_receiver_unittest.cc | 409 const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms; in TEST_F() local
410 const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t); in TEST_F()
430 rtp_header.timestamp += kFrameSizeSamples; in TEST_F()
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| D | audio_coding_module_unittest.cc | 69 const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms; variable
70 const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
167 : rtp_utility_(new RtpData(kFrameSizeSamples, kPayloadType)), in AudioCodingModuleTestOldApi()
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| /external/webrtc/modules/audio_coding/neteq/ |
| D | packet_buffer_unittest.cc | 696 constexpr size_t kFrameSizeSamples = 10; in TEST() local
703 PacketGenerator gen(0, kStartTimeStamp, 0, kFrameSizeSamples); in TEST()
711 .WillRepeatedly(Return(kFrameSizeSamples)); in TEST()
731 EXPECT_EQ(kFrameSizeSamples * 2, in TEST()
736 EXPECT_EQ(kFrameSizeSamples * 2, in TEST()
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