Searched refs:kFrameSizeSamples (Results 1 - 5 of 5) sorted by path
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | audio_coding_module_unittest.cc | 44 const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms; member in namespace:webrtc 45 const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t); 122 : rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)) {
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H A D | audio_coding_module_unittest_oldapi.cc | 43 const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms; member in namespace:webrtc 44 const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t); 122 rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | target_delay_unittest.cc | 47 int16_t audio[kFrameSizeSamples]; 49 for (int n = 0; n < kFrameSizeSamples; ++n) 51 WebRtcPcm16b_Encode(audio, kFrameSizeSamples, payload_); 136 static const int kFrameSizeSamples = 320; // 20 ms @ 16 kHz. member in class:webrtc::TargetDelayTest 144 rtp_info_.header.timestamp += kFrameSizeSamples; 146 ASSERT_EQ(0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2,
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/ |
H A D | audio_classifier_test.cc | 37 const int kFrameSizeSamples = 960; local 50 const int data_size = channels * kFrameSizeSamples;
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/external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
H A D | fine_audio_buffer_unittest.cc | 95 const int kFrameSizeSamples = kSamplesPer10Ms - 50; local 96 RunFineBufferTest(kSampleRate, kFrameSizeSamples); 102 const int kFrameSizeSamples = kSamplesPer10Ms + 50; local 103 RunFineBufferTest(kSampleRate, kFrameSizeSamples);
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