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    Searched refs:kFrameSizeSamples (Results 1 - 6 of 6) sorted by null

  /external/webrtc/webrtc/modules/audio_coding/neteq/test/
neteq_ilbc_quality_test.cc 64 const size_t kFrameSizeSamples = 80; // Samples per 10 ms.
71 in_data + encoded_samples, kFrameSizeSamples),
73 encoded_samples += kFrameSizeSamples;
neteq_pcmu_quality_test.cc 64 const size_t kFrameSizeSamples = 80; // Samples per 10 ms.
71 in_data + encoded_samples, kFrameSizeSamples),
73 encoded_samples += kFrameSizeSamples;
audio_classifier_test.cc 37 const int kFrameSizeSamples = 960;
50 const int data_size = channels * kFrameSizeSamples;
  /external/webrtc/webrtc/modules/audio_coding/test/
target_delay_unittest.cc 46 int16_t audio[kFrameSizeSamples];
48 for (size_t n = 0; n < kFrameSizeSamples; ++n)
50 WebRtcPcm16b_Encode(audio, kFrameSizeSamples, payload_);
135 static const size_t kFrameSizeSamples = 320; // 20 ms @ 16 kHz.
143 rtp_info_.header.timestamp += kFrameSizeSamples;
145 ASSERT_EQ(0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2,
  /external/webrtc/webrtc/modules/audio_device/
fine_audio_buffer_unittest.cc 136 const int kFrameSizeSamples = kSamplesPer10Ms - 50;
137 RunFineBufferTest(kSampleRate, kFrameSizeSamples);
143 const int kFrameSizeSamples = kSamplesPer10Ms + 50;
144 RunFineBufferTest(kSampleRate, kFrameSizeSamples);
  /external/webrtc/webrtc/modules/audio_coding/acm2/
audio_coding_module_unittest_oldapi.cc 55 const int kFrameSizeSamples = kFrameSizeMs / 10 * kNumSamples10ms;
56 const int kPayloadSizeBytes = kFrameSizeSamples * sizeof(int16_t);
158 rtp_utility_(new RtpUtility(kFrameSizeSamples, kPayloadType)),
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