| /frameworks/av/services/audioflinger/ |
| AudioResamplerSinc.h | 42 virtual void resample(int32_t* out, size_t outFrameCount,
50 void resample(int32_t* out, size_t outFrameCount,
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| Android.mk | 77 test-resample.cpp \
88 LOCAL_MODULE:= test-resample
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| test-resample.cpp | 223 resampler->resample((int*) output_vaddr, out_frames, &provider);
224 resampler->resample((int*) output_vaddr, out_frames, &provider);
225 resampler->resample((int*) output_vaddr, out_frames, &provider);
226 resampler->resample((int*) output_vaddr, out_frames, &provider);
243 resampler->resample((int*) output_vaddr, out_frames, &provider);
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| AudioResampler.h | 59 // Resample int16_t samples from provider and accumulate into 'out'.
67 virtual void resample(int32_t* out, size_t outFrameCount,
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| AudioResamplerCubic.h | 34 virtual void resample(int32_t* out, size_t outFrameCount,
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| AudioResamplerCubic.cpp | 35 void AudioResamplerCubic::resample(int32_t* out, size_t outFrameCount, function in class:android::AudioResamplerCubic
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| AudioResampler.cpp | 45 virtual void resample(int32_t* out, size_t outFrameCount,
280 void AudioResamplerOrder1::resample(int32_t* out, size_t outFrameCount,
310 // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
407 // ALOGE("starting resample %d frames, inputIndex=%d, phaseFraction=%d, phaseIncrement=%d",
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| AudioResamplerSinc.cpp | 506 void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, function in class:android::AudioResamplerSinc
531 void AudioResamplerSinc::resample(int32_t* out, size_t outFrameCount, function in class:android::AudioResamplerSinc
[all...] |
| AudioMixer.cpp | 470 case RESAMPLE:
475 ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
695 "Track %d needs downmix + resample", i);
779 // ramp gain - resample to temp buffer and scale/mix in 2nd step
781 // always resample with unity gain when sending to auxiliary buffer to be able
786 t->resampler->resample(temp, outFrameCount, t->bufferProvider);
796 t->resampler->resample(temp, outFrameCount, t->bufferProvider);
803 t->resampler->resample(out, outFrameCount, t->bufferProvider);
[all...] |
| Threads.cpp | [all...] |
| /external/speex/ |
| Android.mk | 46 libspeex/resample.c
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| /external/webrtc/src/common_audio/signal_processing/ |
| Android.mk | 41 resample.c \
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| /frameworks/av/libvideoeditor/vss/src/ |
| VideoEditorResampler.cpp | 162 context->mResampler->resample((int32_t *)pTmpBuffer,
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| /external/webrtc/src/modules/audio_processing/aec/ |
| echo_cancellation.c | 87 int resample; // if the skew is small enough we don't resample member in struct:__anon31381
265 aecpc->resample = kAecFalse;
327 if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
328 // Resample and get a new number of samples
443 aecpc->resample = kAecFalse;
446 aecpc->resample = kAecTrue;
910 if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
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| /frameworks/av/libvideoeditor/lvpp/ |
| VideoEditorSRC.cpp | 141 // Resample to target quality
142 mResampler->resample(pTmpBuffer, kOutputFrameCount, this);
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| /cts/suite/cts/deviceTests/browserbench/assets/octane/ |
| gbemu.js | 426 //Set the buffer passed as our own, as we don't need to resample it:
860 //Resample a chunk of audio:
[all...] |
