/system/media/audio_utils/ |
resampler.c | 18 #define LOG_TAG "resampler" 24 #include <audio_utils/resampler.h> 28 struct resampler { struct 30 SpeexResamplerState *speex_resampler; // handle on speex resampler 41 int32_t speex_delay_ns; // delay introduced by speex resampler in ns 46 // speex based resampler 49 static void resampler_reset(struct resampler_itfe *resampler) 51 struct resampler *rsmp = (struct resampler *)resampler; [all...] |
Android.mk | 11 resampler.c \
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echo_reference.c | 25 #include <audio_utils/resampler.h> 54 void *wr_src_buf; // resampler input buf (either wr_buf or buffer used by write()) 63 struct resampler_itfe *resampler; // input resampler member in struct:echo_reference 64 struct resampler_buffer_provider provider; // resampler buffer provider 124 /* additional space in resampler buffer allowing for extra samples to be returned 125 * by speex resampler when sample rates ratio is not an integer. 163 if (er->resampler != NULL) { 164 er->resampler->reset(er->resampler); [all...] |
/system/media/audio_utils/include/audio_utils/ |
resampler.h | 41 /* call back interface used by the resampler to get new data */ 61 /* resampler interface */ 64 * reset resampler state 66 void (*reset)(struct resampler_itfe *resampler); 71 int (*resample_from_provider)(struct resampler_itfe *resampler, 79 int (*resample_from_input)(struct resampler_itfe *resampler, 85 * return the latency introduced by the resampler in ns. 87 int32_t (*delay_ns)(struct resampler_itfe *resampler); 91 * create a resampler according to input parameters passed. 103 * release resampler resources [all...] |
/external/webkit/Source/WebCore/platform/audio/ |
AudioResamplerKernel.cpp | 40 AudioResamplerKernel::AudioResamplerKernel(AudioResampler* resampler) 41 : m_resampler(resampler)
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AudioBus.cpp | 415 SincResampler resampler(sampleRateRatio); 416 resampler.process(source, destination, sourceLength);
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/frameworks/av/services/audioflinger/ |
AudioMixer.h | 81 // This clears out the resampler's input buffer. 184 AudioResampler* resampler; member in struct:android::AudioMixer::track_t 200 bool doesResample() const { return resampler != NULL; } 201 void resetResampler() { if (resampler != NULL) resampler->reset(); } 203 size_t getUnreleasedFrames() const { return resampler != NULL ? 204 resampler->getUnreleasedFrames() : 0; };
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AudioResampler.cpp | 87 AudioResampler* resampler; local 90 if (property_get("af.resampler.quality", value, 0)) { 101 ALOGV("Create linear Resampler"); 102 resampler = new AudioResamplerOrder1(bitDepth, inChannelCount, sampleRate); 106 ALOGV("Create cubic Resampler"); 107 resampler = new AudioResamplerCubic(bitDepth, inChannelCount, sampleRate); 110 ALOGV("Create sinc Resampler"); 111 resampler = new AudioResamplerSinc(bitDepth, inChannelCount, sampleRate); 116 // initialize resampler 117 resampler->init() [all...] |
AudioMixer.cpp | 122 t->resampler = NULL; 154 delete t->resampler; 196 t->resampler = NULL; 366 // delete the resampler 367 delete track.resampler; 368 track.resampler = NULL; 468 delete track.resampler; 469 track.resampler = NULL; 533 if (value != devSampleRate || resampler != NULL) { 536 if (resampler == NULL) [all...] |
/device/asus/grouper/audio/ |
audio_hw.c | 37 #include <audio_utils/resampler.h> 119 struct resampler_itfe *resampler; member in struct:stream_out 139 struct resampler_itfe *resampler; member in struct:stream_in 214 if (out->resampler) { 215 release_resampler(out->resampler); 216 out->resampler = NULL; 235 if (in->resampler) { 236 release_resampler(in->resampler); 237 in->resampler = NULL; 296 * create a resampler [all...] |
/external/webrtc/src/common_audio/resampler/ |
Android.mk | 20 LOCAL_SRC_FILES := resampler.cc
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/device/ti/panda/audio/ |
audio_hw.c | 45 #include <audio_utils/resampler.h> 606 struct resampler_itfe *resampler; member in struct:omap4_stream_out 625 struct resampler_itfe *resampler; member in struct:omap4_stream_in [all...] |
/external/webrtc/ |
Android.mk | 12 include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/resampler/Android.mk
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/device/samsung/tuna/audio/ |
audio_hw.c | 35 #include <audio_utils/resampler.h> 673 struct resampler_itfe *resampler; member in struct:tuna_stream_out 712 struct resampler_itfe *resampler; member in struct:tuna_stream_in [all...] |
/external/webrtc/src/modules/audio_processing/aec/ |
echo_cancellation.c | 84 void *resampler; member in struct:__anon15277 121 if (WebRtcAec_CreateResampler(&aecpc->resampler) == -1) { 193 WebRtcAec_FreeResampler(aecpc->resampler); 226 if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) { 329 newNrOfSamples = WebRtcAec_ResampleLinear(aecpc->resampler, 434 retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
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/device/samsung/crespo/libaudio/ |
AudioHardware.h | 31 #include <audio_utils/resampler.h>
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