/external/webrtc/webrtc/modules/audio_device/ |
H A D | audio_device_generic.cc | 17 const uint32_t samplesPerSec) { 22 int32_t AudioDeviceGeneric::SetPlayoutSampleRate(const uint32_t samplesPerSec) { argument 16 SetRecordingSampleRate( const uint32_t samplesPerSec) argument
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H A D | audio_device_impl.cc | 1763 int32_t AudioDeviceModuleImpl::SetRecordingSampleRate(const uint32_t samplesPerSec) argument 1767 if (_ptrAudioDevice->SetRecordingSampleRate(samplesPerSec) != 0) 1779 int32_t AudioDeviceModuleImpl::RecordingSampleRate(uint32_t* samplesPerSec) const 1791 *samplesPerSec = sampleRate; 1793 WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: samplesPerSec=%u", *samplesPerSec); 1801 int32_t AudioDeviceModuleImpl::SetPlayoutSampleRate(const uint32_t samplesPerSec) argument 1805 if (_ptrAudioDevice->SetPlayoutSampleRate(samplesPerSec) != 0) 1817 int32_t AudioDeviceModuleImpl::PlayoutSampleRate(uint32_t* samplesPerSec) const 1829 *samplesPerSec [all...] |
/external/webrtc/webrtc/modules/audio_device/test/ |
H A D | func_test_manager.h | 53 uint32_t samplesPerSec; member in struct:AudioPacket 92 const uint32_t samplesPerSec, 102 const uint32_t samplesPerSec,
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H A D | func_test_manager.cc | 198 const uint32_t samplesPerSec, 212 packet->samplesPerSec = samplesPerSec; 343 const uint32_t samplesPerSec, 369 const uint32_t samplesPerSecIn = packet->samplesPerSec; 373 int32_t fsOutHz(samplesPerSec); 418 samplesPerSecIn, samplesPerSec); 456 samplesPerSecIn, samplesPerSec); 1239 uint32_t samplesPerSec(0); 1257 EXPECT_EQ(0, audioDevice->PlayoutSampleRate(&samplesPerSec)); 193 RecordedDataIsAvailable( const void* audioSamples, const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, const uint32_t totalDelayMS, const int32_t clockDrift, const uint32_t currentMicLevel, const bool keyPressed, uint32_t& newMicLevel) argument 339 NeedMorePlayData( const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument [all...] |
/external/webrtc/webrtc/modules/audio_device/include/ |
H A D | fake_audio_device.h | 134 virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) { argument 137 virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const { 140 virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) { argument 143 virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const { return 0; }
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/external/webrtc/webrtc/modules/media_file/ |
H A D | media_file_utility.cc | 255 int32_t ModuleFileUtility::InitWavCodec(uint32_t samplesPerSec, argument 261 codec_info_.plfreq = samplesPerSec; 263 codec_info_.rate = bitsPerSample * samplesPerSec; 282 if(samplesPerSec == 8000) 287 else if(samplesPerSec == 16000) 292 else if(samplesPerSec == 32000) 299 else if(samplesPerSec == 11025) 306 else if(samplesPerSec == 22050) 313 else if(samplesPerSec == 44100) 320 else if(samplesPerSec [all...] |
/external/webrtc/webrtc/voice_engine/ |
H A D | voe_base_impl.cc | 86 const uint32_t samplesPerSec, 93 nullptr, 0, audioSamples, samplesPerSec, nChannels, nSamples, 101 const uint32_t samplesPerSec, 106 GetPlayoutData(static_cast<int>(samplesPerSec), nChannels, nSamples, true, 82 RecordedDataIsAvailable(const void* audioSamples, const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, const uint32_t totalDelayMS, const int32_t clockDrift, const uint32_t currentMicLevel, const bool keyPressed, uint32_t& newMicLevel) argument 98 NeedMorePlayData(const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument
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H A D | transmit_mixer.cc | 323 uint32_t samplesPerSec, 331 "nChannels=%" PRIuS ", samplesPerSec=%u, totalDelayMS=%u, " 333 nSamples, nChannels, samplesPerSec, totalDelayMS, clockDrift, 340 samplesPerSec); local 320 PrepareDemux(const void* audioSamples, size_t nSamples, size_t nChannels, uint32_t samplesPerSec, uint16_t totalDelayMS, int32_t clockDrift, uint16_t currentMicLevel, bool keyPressed) argument
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/external/webrtc/webrtc/modules/audio_device/android/ |
H A D | audio_device_unittest.cc | 387 const uint32_t samplesPerSec, 397 const uint32_t samplesPerSec, 427 const uint32_t samplesPerSec, 449 const uint32_t samplesPerSec, 423 RealRecordedDataIsAvailable(const void* audioSamples, const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, const uint32_t totalDelayMS, const int32_t clockDrift, const uint32_t currentMicLevel, const bool keyPressed, uint32_t& newMicLevel) argument 446 RealNeedMorePlayData(const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument
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/external/webrtc/webrtc/modules/audio_device/ios/ |
H A D | audio_device_unittest_ios.cc | 377 const uint32_t samplesPerSec, 387 const uint32_t samplesPerSec, 417 const uint32_t samplesPerSec, 441 const uint32_t samplesPerSec, 413 RealRecordedDataIsAvailable(const void* audioSamples, const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, const uint32_t totalDelayMS, const int32_t clockDrift, const uint32_t currentMicLevel, const bool keyPressed, uint32_t& newMicLevel) argument 438 RealNeedMorePlayData(const size_t nSamples, const size_t nBytesPerSample, const size_t nChannels, const uint32_t samplesPerSec, void* audioSamples, size_t& nSamplesOut, int64_t* elapsed_time_ms, int64_t* ntp_time_ms) argument
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