| /external/chromium_org/third_party/webrtc/modules/audio_device/test/ |
| D | func_test_manager.cc | 148 const uint32_t samplesPerSec, in RecordedDataIsAvailable() argument
162 packet->samplesPerSec = samplesPerSec; in RecordedDataIsAvailable()
293 const uint32_t samplesPerSec, in NeedMorePlayData() argument
319 const uint32_t samplesPerSecIn = packet->samplesPerSec; in NeedMorePlayData()
324 int32_t fsOutHz(samplesPerSec); in NeedMorePlayData()
371 samplesPerSecIn, samplesPerSec); in NeedMorePlayData()
413 samplesPerSecIn, samplesPerSec); in NeedMorePlayData()
1214 uint32_t samplesPerSec(0); in TestAudioTransport() local
1232 EXPECT_EQ(0, audioDevice->PlayoutSampleRate(&samplesPerSec)); in TestAudioTransport()
1233 if (samplesPerSec == 48000) { in TestAudioTransport()
[all …]
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| D | func_test_manager.h | 69 uint32_t samplesPerSec; member
109 const uint32_t samplesPerSec,
119 const uint32_t samplesPerSec,
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| /external/chromium_org/third_party/webrtc/modules/audio_device/ |
| D | audio_device_generic.cc | 17 const uint32_t samplesPerSec) in SetRecordingSampleRate() argument
25 const uint32_t samplesPerSec) in SetPlayoutSampleRate() argument
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| D | audio_device_impl.h | 189 const uint32_t samplesPerSec) OVERRIDE;
191 uint32_t* samplesPerSec) const OVERRIDE;
193 const uint32_t samplesPerSec) OVERRIDE;
195 uint32_t* samplesPerSec) const OVERRIDE;
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| D | audio_device_impl.cc | 1856 int32_t AudioDeviceModuleImpl::SetRecordingSampleRate(const uint32_t samplesPerSec) in SetRecordingSampleRate() argument
1860 if (_ptrAudioDevice->SetRecordingSampleRate(samplesPerSec) != 0) in SetRecordingSampleRate()
1872 int32_t AudioDeviceModuleImpl::RecordingSampleRate(uint32_t* samplesPerSec) const in RecordingSampleRate()
1884 *samplesPerSec = sampleRate; in RecordingSampleRate()
1886 … WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: samplesPerSec=%u", *samplesPerSec); in RecordingSampleRate()
1894 int32_t AudioDeviceModuleImpl::SetPlayoutSampleRate(const uint32_t samplesPerSec) in SetPlayoutSampleRate() argument
1898 if (_ptrAudioDevice->SetPlayoutSampleRate(samplesPerSec) != 0) in SetPlayoutSampleRate()
1910 int32_t AudioDeviceModuleImpl::PlayoutSampleRate(uint32_t* samplesPerSec) const in PlayoutSampleRate()
1922 *samplesPerSec = sampleRate; in PlayoutSampleRate()
1924 … WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: samplesPerSec=%u", *samplesPerSec); in PlayoutSampleRate()
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| D | audio_device_generic.h | 141 const uint32_t samplesPerSec);
143 const uint32_t samplesPerSec);
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| /external/chromium_org/third_party/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 | 175 virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) = 0;
176 virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const = 0;
177 virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) = 0;
178 virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const = 0;
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| D | audio_device_defines.h | 61 const uint32_t samplesPerSec,
71 const uint32_t samplesPerSec,
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| /external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
| D | audio_device_template.h | 390 const uint32_t samplesPerSec) { in SetRecordingSampleRate() argument
391 return input_.SetRecordingSampleRate(samplesPerSec); in SetRecordingSampleRate()
395 const uint32_t samplesPerSec) { in SetPlayoutSampleRate() argument
396 return output_.SetPlayoutSampleRate(samplesPerSec); in SetPlayoutSampleRate()
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| D | audio_record_jni.cc | 778 int32_t AudioRecordJni::SetRecordingSampleRate(const uint32_t samplesPerSec) { in SetRecordingSampleRate() argument
779 if (samplesPerSec > 48000 || samplesPerSec < 8000) in SetRecordingSampleRate()
787 if (samplesPerSec == 44100) in SetRecordingSampleRate()
793 _samplingFreqIn = samplesPerSec / 1000; in SetRecordingSampleRate()
797 _ptrAudioBuffer->SetRecordingSampleRate(samplesPerSec); in SetRecordingSampleRate()
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| D | opensles_common.cc | 29 configuration.samplesPerSec = sample_rate * 1000; in CreatePcmConfiguration()
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| D | audio_track_jni.cc | 857 int32_t AudioTrackJni::SetPlayoutSampleRate(const uint32_t samplesPerSec) { in SetPlayoutSampleRate() argument
858 if (samplesPerSec > 48000 || samplesPerSec < 8000) in SetPlayoutSampleRate()
866 if (samplesPerSec == 44100) in SetPlayoutSampleRate()
872 _samplingFreqOut = samplesPerSec / 1000; in SetPlayoutSampleRate()
876 _ptrAudioBuffer->SetPlayoutSampleRate(samplesPerSec); in SetPlayoutSampleRate()
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| D | audio_record_jni.h | 110 int32_t SetRecordingSampleRate(const uint32_t samplesPerSec);
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| D | audio_track_jni.h | 97 int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec);
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| /external/chromium_org/third_party/webrtc/voice_engine/ |
| D | voe_base_impl.h | 70 uint32_t samplesPerSec,
80 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 | 125 uint32_t samplesPerSec, in RecordedDataIsAvailable() argument
136 nSamples, nBytesPerSample, nChannels, samplesPerSec, in RecordedDataIsAvailable()
139 NULL, 0, audioSamples, samplesPerSec, nChannels, nSamples, in RecordedDataIsAvailable()
149 uint32_t samplesPerSec, in NeedMorePlayData() argument
158 nSamples, nBytesPerSample, nChannels, samplesPerSec); in NeedMorePlayData()
160 GetPlayoutData(static_cast<int>(samplesPerSec), in NeedMorePlayData()
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| D | transmit_mixer.cc | 323 uint32_t samplesPerSec, in PrepareDemux() argument
332 "currentMicLevel=%u)", nSamples, nChannels, samplesPerSec, in PrepareDemux()
339 samplesPerSec); in PrepareDemux()
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| /external/chromium_org/third_party/libjingle/source/talk/app/webrtc/test/ |
| D | fakeaudiocapturemodule_unittest.cc | 62 const uint32_t samplesPerSec, in RecordedDataIsAvailable() argument
85 const uint32_t samplesPerSec, in NeedMorePlayData() argument
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| /external/chromium_org/third_party/webrtc/modules/media_file/source/ |
| D | media_file_utility.cc | 633 int32_t ModuleFileUtility::InitWavCodec(uint32_t samplesPerSec, in InitWavCodec() argument
639 codec_info_.plfreq = samplesPerSec; in InitWavCodec()
641 codec_info_.rate = bitsPerSample * samplesPerSec; in InitWavCodec()
660 if(samplesPerSec == 8000) in InitWavCodec()
665 else if(samplesPerSec == 16000) in InitWavCodec()
670 else if(samplesPerSec == 32000) in InitWavCodec()
677 else if(samplesPerSec == 11025) in InitWavCodec()
684 else if(samplesPerSec == 22050) in InitWavCodec()
691 else if(samplesPerSec == 44100) in InitWavCodec()
698 else if(samplesPerSec == 48000) in InitWavCodec()
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| D | media_file_utility.h | 238 int32_t InitWavCodec(uint32_t samplesPerSec,
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| /external/qemu/distrib/sdl-1.2.15/src/audio/ums/ |
| D | SDL_umsaudio.c | 232 long samplesPerSec; in UMS_OpenAudio() local
326 samplesPerSec = this->hidden->bytesPerSample * outRate; in UMS_OpenAudio()
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| /external/chromium_org/media/audio/android/ |
| D | opensles_input.cc | 37 format_.samplesPerSec = static_cast<SLuint32>(params.sample_rate() * 1000); in OpenSLESInputStream()
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| D | opensles_output.cc | 40 format_.samplesPerSec = static_cast<SLuint32>(params.sample_rate() * 1000); in OpenSLESOutputStream()
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