Searched refs:kFrameSizeSamples (Results 1 - 5 of 5) sorted by path

/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/
H A Daudio_coding_module_unittest.cc44 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)) {
H A Daudio_coding_module_unittest_oldapi.cc43 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)),
/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/
H A Dtarget_delay_unittest.cc47 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,
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/test/
H A Daudio_classifier_test.cc37 const int kFrameSizeSamples = 960; local
50 const int data_size = channels * kFrameSizeSamples;
/external/chromium_org/third_party/webrtc/modules/audio_device/android/
H A Dfine_audio_buffer_unittest.cc95 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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