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Searched refs:kSampleRateHz (Results 1 – 25 of 27) sorted by relevance

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/external/webrtc/webrtc/modules/audio_processing/vad/
Dvoice_activity_detector_unittest.cc53 const int kSampleRateHz = 16000; in TEST() local
54 const int kLength10Ms = kSampleRateHz / 100; in TEST()
68 ASSERT_EQ(fseek(pcm_file, kStartTimeSec * kSampleRateHz * sizeof(data[0]), in TEST()
75 vad.ProcessChunk(&data[0], data.size(), kSampleRateHz); in TEST()
88 const int kSampleRateHz = 32000; in TEST() local
89 const int kLength10Ms = kSampleRateHz / 100; in TEST()
103 ASSERT_EQ(fseek(pcm_file, kStartTimeSec * kSampleRateHz * sizeof(data[0]), in TEST()
110 vad.ProcessChunk(&data[0], data.size(), kSampleRateHz); in TEST()
133 vad.ProcessChunk(&data[0], data.size(), kSampleRateHz); in TEST()
156 vad.ProcessChunk(&data[0], data.size(), 2 * kSampleRateHz); in TEST()
Dcommon.h14 static const int kSampleRateHz = 16000; variable
15 static const size_t kLength10Ms = kSampleRateHz / 100;
Dvad_audio_proc.h53 static_cast<size_t>(kSampleRateHz / 200);
61 static_cast<size_t>(kSampleRateHz / 100);
Dstandalone_vad.cc67 assert(WebRtcVad_ValidRateAndFrameLength(kSampleRateHz, index_) == 0); in GetActivity()
69 int activity = WebRtcVad_Process(vad_, kSampleRateHz, buffer_, index_); in GetActivity()
Dvoice_activity_detector.cc44 if (sample_rate_hz != kSampleRateHz) { in ProcessChunk()
46 resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels), in ProcessChunk()
Dvad_audio_proc.cc36 kSampleRateHz / static_cast<float>(VadAudioProc::kDftSize);
260 kSampleRateHz / 2, gains, lags, kNumPitchSubframes, kNum10msSubframes, in PitchAnalysis()
/external/webrtc/webrtc/modules/audio_coding/neteq/
Dneteq_impl_unittest.cc416 const int kSampleRateHz = 8000; in TEST_F() local
418 static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms. in TEST_F()
459 "dummy name", kPayloadType, kSampleRateHz)); in TEST_F()
466 const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000); in TEST_F()
510 const int kSampleRateHz = 8000; in TEST_F() local
512 static_cast<size_t>(10 * kSampleRateHz / 1000); // 10 ms. in TEST_F()
531 kSampleRateHz, _, _)) in TEST_F()
538 "dummy name", kPayloadType, kSampleRateHz)); in TEST_F()
545 const size_t kMaxOutputSize = static_cast<size_t>(10 * kSampleRateHz / 1000); in TEST_F()
574 kSampleRateHz, _, _)) in TEST_F()
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Dnack_unittest.cc26 const int kSampleRateHz = 16000; variable
59 nack->UpdateSampleRate(kSampleRateHz); in TEST()
77 nack->UpdateSampleRate(kSampleRateHz); in TEST()
106 nack->UpdateSampleRate(kSampleRateHz); in TEST()
155 nack->UpdateSampleRate(kSampleRateHz); in TEST()
217 nack->UpdateSampleRate(kSampleRateHz); in TEST()
288 nack->UpdateSampleRate(kSampleRateHz); in TEST()
339 nack->UpdateSampleRate(kSampleRateHz); in TEST()
366 nack->UpdateSampleRate(kSampleRateHz); in TEST()
390 nack->UpdateSampleRate(kSampleRateHz); in TEST()
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/external/webrtc/webrtc/modules/audio_coding/codecs/ilbc/
Daudio_encoder_ilbc.cc23 const int kSampleRateHz = 8000; variable
40 static_cast<size_t>(kSampleRateHz / 100 * (frame_size_ms / 10)) <= in IsOk()
64 return kSampleRateHz; in SampleRateHz()
104 RTC_DCHECK_EQ(static_cast<size_t>(kSampleRateHz / 100), audio.size()); in EncodeInternal()
106 input_buffer_ + kSampleRateHz / 100 * num_10ms_frames_buffered_); in EncodeInternal()
120 kSampleRateHz / 100 * num_10ms_frames_per_packet_, in EncodeInternal()
/external/webrtc/webrtc/modules/audio_coding/codecs/g722/
Daudio_encoder_g722.cc22 const size_t kSampleRateHz = 16000; variable
50 kSampleRateHz / 100 * num_10ms_frames_per_packet_; in AudioEncoderG722()
68 return kSampleRateHz; in SampleRateHz()
78 return kSampleRateHz / 2; in RtpTimestampRateHz()
105 const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_; in EncodeInternal()
106 for (size_t i = 0; i < kSampleRateHz / 100; ++i) in EncodeInternal()
161 return kSampleRateHz / 100 * num_10ms_frames_per_packet_; in SamplesPerChannel()
/external/webrtc/webrtc/modules/audio_processing/aec/
Dsystem_delay_unittest.cc82 static const int kSampleRateHz[] = {8000, 16000}; variable
84 sizeof(kSampleRateHz) / sizeof(*kSampleRateHz);
206 Init(kSampleRateHz[i]); in TEST_F()
234 Init(kSampleRateHz[i]); in TEST_F()
269 Init(kSampleRateHz[i]); in TEST_F()
319 Init(kSampleRateHz[i]); in TEST_F()
380 Init(kSampleRateHz[i]); in TEST_F()
410 Init(kSampleRateHz[i]); in TEST_F()
454 Init(kSampleRateHz[i]); in TEST_F()
515 Init(kSampleRateHz[i]); in TEST_F()
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/external/webrtc/webrtc/modules/audio_processing/beamformer/
Dnonlinear_beamformer_unittest.cc24 const int kSampleRateHz = 16000; variable
58 bf.Initialize(kChunkSizeMs, kSampleRateHz); in TEST()
75 bf.Initialize(kChunkSizeMs, kSampleRateHz); in TEST()
96 bf.Initialize(kChunkSizeMs, kSampleRateHz); in TEST()
115 bf.Initialize(kChunkSizeMs, kSampleRateHz); in TEST()
134 bf.Initialize(kChunkSizeMs, kSampleRateHz); in TEST()
/external/webrtc/webrtc/test/fuzzers/
Daudio_decoder_ilbc_fuzzer.cc17 static const int kSampleRateHz = 8000; in FuzzOneInput() local
18 static const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10; in FuzzOneInput()
20 FuzzAudioDecoder(data, size, &dec, kSampleRateHz, sizeof(output), output); in FuzzOneInput()
Daudio_decoder_opus_fuzzer.cc18 const int kSampleRateHz = 48000; in FuzzOneInput() local
19 const size_t kAllocatedOuputSizeSamples = kSampleRateHz / 10; // 100 ms. in FuzzOneInput()
21 FuzzAudioDecoder(data, size, &dec, kSampleRateHz, sizeof(output), output); in FuzzOneInput()
Daudio_decoder_isacfix_fuzzer.cc17 static const int kSampleRateHz = 16000; in FuzzOneInput() local
20 FuzzAudioDecoder(data, size, &dec, kSampleRateHz, sizeof(output), output); in FuzzOneInput()
/external/webrtc/webrtc/modules/audio_coding/acm2/
Daudio_coding_module_unittest_oldapi.cc52 const int kSampleRateHz = 16000; variable
53 const int kNumSamples10ms = kSampleRateHz / 100;
76 rtp_header->header.payload_type_frequency = kSampleRateHz; in Populate()
170 input_frame_.sample_rate_hz_ = kSampleRateHz; in SetUp()
172 input_frame_.samples_per_channel_ = kSampleRateHz * 10 / 1000; // 10 ms. in SetUp()
173 static_assert(kSampleRateHz * 10 / 1000 <= AudioFrame::kMaxDataSizeSamples, in SetUp()
186 ASSERT_EQ(0, AudioCodingModule::Codec("L16", &codec_, kSampleRateHz, 1)); in SetUpL16Codec()
299 const int kSampleRateHz = 32000; in TEST_F() local
300 EXPECT_EQ(0, acm_->PlayoutData10Ms(kSampleRateHz, &audio_frame)); in TEST_F()
304 EXPECT_EQ(static_cast<size_t>(kSampleRateHz / 100), in TEST_F()
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Drent_a_codec_unittest.cc105 const int kSampleRateHz = 8000; in TEST() local
108 .WillRepeatedly(Return(kSampleRateHz)); in TEST()
116 const int kPacketSizeSamples = kSampleRateHz / 100; in TEST()
/external/webrtc/webrtc/voice_engine/test/auto_test/standard/
Dfile_before_streaming_test.cc16 const int kSampleRateHz = 16000; variable
46 for (int i = 0; i < kSampleRateHz / 1000 * (kTestDurationMs + 1000); i++) { in GenerateInputFile()
68 kSampleRateHz / 1000 * kSkipOutputMs, SEEK_SET)); in VerifyOutput()
76 ASSERT_GE((samples_read * 1000.0) / kSampleRateHz, 0.4 * kTestDurationMs); in VerifyOutput()
/external/webrtc/webrtc/modules/audio_coding/codecs/g711/
Daudio_encoder_pcm.h78 : AudioEncoderPcm(config, kSampleRateHz) {} in AudioEncoderPcmA()
89 static const int kSampleRateHz = 8000;
100 : AudioEncoderPcm(config, kSampleRateHz) {} in AudioEncoderPcmU()
111 static const int kSampleRateHz = 8000;
/external/webrtc/webrtc/modules/audio_conference_mixer/test/
Daudio_conference_mixer_unittest.cc109 const int kSampleRateHz = 32000; in TEST() local
121 participants[i].fake_frame()->sample_rate_hz_ = kSampleRateHz; in TEST()
127 participants[i].fake_frame()->samples_per_channel_ = kSampleRateHz / 100; in TEST()
137 .WillRepeatedly(Return(kSampleRateHz)); in TEST()
/external/webrtc/webrtc/modules/utility/source/
Dfile_player_unittests.cc32 static const int kSampleRateHz = 8000; member in webrtc::FilePlayerTest
63 int16_t out[10 * kSampleRateHz / 1000] = {0}; in PlayFileAndCheck()
66 player_->Get10msAudioFromFile(out, num_samples, kSampleRateHz)); in PlayFileAndCheck()
/external/webrtc/webrtc/modules/audio_coding/test/
Dtarget_delay_unittest.cc33 ASSERT_EQ(0, AudioCodingModule::Codec("L16", &codec, kSampleRateHz, 1)); in SetUp()
133 static const int kSampleRateHz = 16000; member in webrtc::TargetDelayTest
155 ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_); in Pull()
157 ASSERT_TRUE(kSampleRateHz / 100 == frame.samples_per_channel_); in Pull()
/external/webrtc/webrtc/modules/audio_coding/codecs/opus/
Daudio_encoder_opus.cc22 const int kSampleRateHz = 48000; variable
114 return kSampleRateHz; in SampleRateHz()
220 return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels; in SamplesPer10msFrame()
/external/webrtc/webrtc/modules/audio_processing/
Dsplitting_filter_unittest.cc37 static const int kSampleRateHz = 48000; in TEST() local
60 (i * kSamplesPer48kHzChannel + k) / kSampleRateHz); in TEST()
/external/webrtc/webrtc/tools/agc/
Dactivity_metric.cc100 kSampleRateHz / 100 || in AddAudio()
101 frame.sample_rate_hz_ != kSampleRateHz) in AddAudio()

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