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  /external/webrtc/webrtc/common_audio/signal_processing/
resample_48khz.c 13 * This file contains resampling functions between 48 kHz and nb/wb.
resample_by_2.c 13 * This file contains the resampling by two functions.
resample_by_2_mips.c 13 * This file contains the resampling by two functions.
  /external/webrtc/webrtc/modules/audio_processing/aec/
aec_resampler.c 12 * clock skew by resampling the farend signal.
  /frameworks/base/media/java/android/media/
PlaybackParams.java 53 * <li> <em>Pitch equals speed.</em> Speed change will be done by <em>resampling</em>,
SyncParams.java 38 * Change the speed of audio playback by resampling the audio.</li>
  /external/libvpx/libvpx/
y4minput.c 183 We use a resampling filter to shift the site locations one quarter pixel (at
187 mpeg2 and jpeg cases (thus requiring no vertical resampling).*/
273 We use a resampling filter to shift the site locations one quarter pixel (at
429 We use a resampling filter to decimate the chroma planes by two in the
492 We use a resampling filter to shift the original site locations one quarter
494 Then we use a second resampling filter to decimate the chroma planes by two
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  /hardware/intel/common/omx-components/videocodec/libvpx_internal/libvpx/
y4minput.c 183 We use a resampling filter to shift the site locations one quarter pixel (at
187 mpeg2 and jpeg cases (thus requiring no vertical resampling).*/
273 We use a resampling filter to shift the site locations one quarter pixel (at
429 We use a resampling filter to decimate the chroma planes by two in the
492 We use a resampling filter to shift the original site locations one quarter
494 Then we use a second resampling filter to decimate the chroma planes by two
    [all...]
CHANGELOG 166 with spatial resampling enabled.
402 vp8e_get_preview fixed when spatial resampling enabled
  /frameworks/av/media/libstagefright/codecs/m4v_h263/enc/src/
mp4lib_int.h 114 Int horSamp_n; /* VOL hor. resampling of ref. VOL given by */
116 Int verSamp_n; /* VOL ver. resampling of ref. VOL given by */
  /frameworks/av/services/audioflinger/
AudioResamplerFirProcess.h 246 /* Calculates a single output frame from a polyphase resampling filter.
263 * Calculates a single output frame from a polyphase resampling filter,
  /external/opencv3/3rdparty/include/ffmpeg_/libavutil/
avutil.h 42 * @li @ref lswr "libswresample" audio resampling, format conversion and mixing
  /external/webrtc/webrtc/modules/audio_coding/acm2/
audio_coding_module_impl.h 224 // Preprocessing of input audio, including resampling and down-mixing if
acm_receiver.h 78 // audio. If set -1 indicates to resampling is
  /external/opencv3/3rdparty/libjpeg/
change.log 123 common 2x2 chroma subsampling case without additional spatial resampling.
124 Separate spatial resampling for those kind of files is now only necessary
128 spatial resampling.
jdsample.c 17 * An excellent reference for image resampling is
  /external/speex/libspeex/
resample.c 5 Arbitrary resampling code
45 This algorithm is based on this original resampling algorithm:
46 Smith, Julius O. Digital Audio Resampling Home Page
222 close to the Nyquist rate (from an A/D or a previous resampling filter)
587 /* Choose the resampling type that requires the least amount of memory */
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  /hardware/libhardware/modules/audio_remote_submix/
audio_hw.cpp 88 // Whether resampling is enabled.
635 ALOGE("out_set_sample_rate() resampling enabled can't change sample rate from "
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  /external/libpng/contrib/tools/
genpng.c 196 * This is the equivalent of aliased rendering or resampling an image with
204 * This is the equivalent of 'pixel coverage' alpha calculation or resampling
210 * image is 0.5 per sample, but the resampling reduces this.
  /frameworks/native/libs/input/
InputTransport.cpp 19 // Log debug messages about touch event resampling
47 // Latency added during resampling. A few milliseconds doesn't hurt much but
720 // Find the data to use for resampling.
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  /external/webrtc/webrtc/modules/audio_coding/test/
opus_test.cc 259 // If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
  /external/webrtc/webrtc/modules/audio_device/ios/
audio_device_ios.h 171 // audio. The selected stream format is selected to avoid internal resampling
  /external/libopus/silk/
enc_API.c 254 /* Input buffering/resampling and encoding */
291 /* Combine left and right channels before resampling */
  /external/webrtc/webrtc/common_audio/signal_processing/include/
signal_processing_library.h 749 * RESAMPLING FUNCTIONS AND THEIR STRUCTS ARE DEFINED BELOW
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  /external/ImageMagick/ImageMagick/api/
distort.html 152 <dd> o 'filter' Set filter to use for area-resampling (scale shrinking). Set to 'point' to turn off and use 'interpolate' lookup instead </dd>

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