/external/webrtc/webrtc/modules/audio_device/test/ |
D | func_test_manager.cc | 198 const uint32_t samplesPerSec, in RecordedDataIsAvailable() argument 212 packet->samplesPerSec = samplesPerSec; in RecordedDataIsAvailable() 343 const uint32_t samplesPerSec, in NeedMorePlayData() argument 369 const uint32_t samplesPerSecIn = packet->samplesPerSec; in NeedMorePlayData() 373 int32_t fsOutHz(samplesPerSec); in NeedMorePlayData() 418 samplesPerSecIn, samplesPerSec); in NeedMorePlayData() 456 samplesPerSecIn, samplesPerSec); in NeedMorePlayData() 1239 uint32_t samplesPerSec(0); in TestAudioTransport() local 1257 EXPECT_EQ(0, audioDevice->PlayoutSampleRate(&samplesPerSec)); in TestAudioTransport() 1258 if (samplesPerSec == 48000) { in TestAudioTransport() [all …]
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D | func_test_manager.h | 53 uint32_t samplesPerSec; member 92 const uint32_t samplesPerSec, 102 const uint32_t samplesPerSec,
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/external/webrtc/webrtc/modules/audio_device/android/ |
D | opensles_player.cc | 201 format.samplesPerSec = SL_SAMPLINGRATE_8; in CreatePCMConfiguration() 204 format.samplesPerSec = SL_SAMPLINGRATE_16; in CreatePCMConfiguration() 207 format.samplesPerSec = SL_SAMPLINGRATE_22_05; in CreatePCMConfiguration() 210 format.samplesPerSec = SL_SAMPLINGRATE_32; in CreatePCMConfiguration() 213 format.samplesPerSec = SL_SAMPLINGRATE_44_1; in CreatePCMConfiguration() 216 format.samplesPerSec = SL_SAMPLINGRATE_48; in CreatePCMConfiguration()
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D | opensles_common.cc | 29 configuration.samplesPerSec = sample_rate * 1000; in CreatePcmConfiguration()
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D | audio_device_unittest.cc | 387 const uint32_t samplesPerSec, 397 const uint32_t samplesPerSec, 427 const uint32_t samplesPerSec, in RealRecordedDataIsAvailable() argument 449 const uint32_t samplesPerSec, in RealNeedMorePlayData() argument
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D | audio_device_template.h | 437 int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) override { in SetPlayoutSampleRate() argument
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/external/webrtc/webrtc/modules/audio_device/include/ |
D | fake_audio_device.h | 134 virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) { in SetRecordingSampleRate() argument 137 virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const { in RecordingSampleRate() argument 140 virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) { in SetPlayoutSampleRate() argument 143 virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const { return 0; } in PlayoutSampleRate() argument
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D | audio_device.h | 177 virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) = 0; 178 virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const = 0; 179 virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) = 0; 180 virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const = 0;
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D | audio_device_defines.h | 53 const uint32_t samplesPerSec, 63 const uint32_t samplesPerSec,
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/external/webrtc/webrtc/modules/audio_device/ |
D | audio_device_impl.h | 175 int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) override; 176 int32_t RecordingSampleRate(uint32_t* samplesPerSec) const override; 177 int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) override; 178 int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const override;
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D | audio_device_generic.cc | 17 const uint32_t samplesPerSec) { in SetRecordingSampleRate() argument 22 int32_t AudioDeviceGeneric::SetPlayoutSampleRate(const uint32_t samplesPerSec) { in SetPlayoutSampleRate() argument
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D | audio_device_impl.cc | 1763 int32_t AudioDeviceModuleImpl::SetRecordingSampleRate(const uint32_t samplesPerSec) in SetRecordingSampleRate() argument 1767 if (_ptrAudioDevice->SetRecordingSampleRate(samplesPerSec) != 0) in SetRecordingSampleRate() 1779 int32_t AudioDeviceModuleImpl::RecordingSampleRate(uint32_t* samplesPerSec) const in RecordingSampleRate() 1791 *samplesPerSec = sampleRate; in RecordingSampleRate() 1793 … WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: samplesPerSec=%u", *samplesPerSec); in RecordingSampleRate() 1801 int32_t AudioDeviceModuleImpl::SetPlayoutSampleRate(const uint32_t samplesPerSec) in SetPlayoutSampleRate() argument 1805 if (_ptrAudioDevice->SetPlayoutSampleRate(samplesPerSec) != 0) in SetPlayoutSampleRate() 1817 int32_t AudioDeviceModuleImpl::PlayoutSampleRate(uint32_t* samplesPerSec) const in PlayoutSampleRate() 1829 *samplesPerSec = sampleRate; in PlayoutSampleRate() 1831 … WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: samplesPerSec=%u", *samplesPerSec); in PlayoutSampleRate()
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D | audio_device_generic.h | 131 virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec); 132 virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec);
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/external/webrtc/webrtc/modules/media_file/ |
D | media_file_utility.cc | 255 int32_t ModuleFileUtility::InitWavCodec(uint32_t samplesPerSec, in InitWavCodec() argument 261 codec_info_.plfreq = samplesPerSec; in InitWavCodec() 263 codec_info_.rate = bitsPerSample * samplesPerSec; in InitWavCodec() 282 if(samplesPerSec == 8000) in InitWavCodec() 287 else if(samplesPerSec == 16000) in InitWavCodec() 292 else if(samplesPerSec == 32000) in InitWavCodec() 299 else if(samplesPerSec == 11025) in InitWavCodec() 306 else if(samplesPerSec == 22050) in InitWavCodec() 313 else if(samplesPerSec == 44100) in InitWavCodec() 320 else if(samplesPerSec == 48000) in InitWavCodec()
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D | media_file_utility.h | 182 int32_t InitWavCodec(uint32_t samplesPerSec,
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/external/webrtc/webrtc/voice_engine/ |
D | voe_base_impl.h | 61 const uint32_t samplesPerSec, 70 const uint32_t samplesPerSec,
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D | transmit_mixer.h | 56 uint32_t samplesPerSec, 178 int samplesPerSec);
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D | voe_base_impl.cc | 86 const uint32_t samplesPerSec, in RecordedDataIsAvailable() argument 93 nullptr, 0, audioSamples, samplesPerSec, nChannels, nSamples, in RecordedDataIsAvailable() 101 const uint32_t samplesPerSec, in NeedMorePlayData() argument 106 GetPlayoutData(static_cast<int>(samplesPerSec), nChannels, nSamples, true, in NeedMorePlayData()
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D | transmit_mixer.cc | 323 uint32_t samplesPerSec, in PrepareDemux() argument 333 nSamples, nChannels, samplesPerSec, totalDelayMS, clockDrift, in PrepareDemux() 340 samplesPerSec); in PrepareDemux()
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/external/webrtc/talk/app/webrtc/test/ |
D | fakeaudiocapturemodule_unittest.cc | 62 const uint32_t samplesPerSec, in RecordedDataIsAvailable() argument 86 const uint32_t samplesPerSec, in NeedMorePlayData() argument
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/external/webrtc/webrtc/modules/audio_device/ios/ |
D | audio_device_unittest_ios.cc | 377 const uint32_t samplesPerSec, 387 const uint32_t samplesPerSec, 417 const uint32_t samplesPerSec, in RealRecordedDataIsAvailable() argument 441 const uint32_t samplesPerSec, in RealNeedMorePlayData() argument
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/external/walt/android/WALT/app/src/main/jni/ |
D | player.c | 258 format_pcm.samplesPerSec = (SLuint32) optimalFrameRate * 1000; in Java_org_chromium_latency_walt_AudioTest_createBufferQueueAudioPlayer() 409 format_pcm.samplesPerSec = (SLuint32) optimalFrameRate * 1000; in Java_org_chromium_latency_walt_AudioTest_createAudioRecorder()
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/external/drrickorang/LoopbackApp/app/src/main/cpp/ |
D | sles.cpp | 647 pcm.samplesPerSec = pSles->sampleRate * 1000; in slesCreateServer()
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